• The Journal of the Acoustical Society of Korea
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• v.29 no.8
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• pp.504-508
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• 2010

In conventional diagnostic ultrasound strain imaging, when displaying strain image on a monitor, human visual characteristics are utilized such that hard regions are displayed as dark and soft regions are displayed as bright. Thus, hard regions representing tumor or cancer are displayed as dark, decreasing the contrast inside the lesion. Because the lesion area is stiff and thus displayed as dark, a method of inverting the image brightness and thereby increasing the contrast in the lesion for better diagnostic purposes is proposed wherein a postcompression signal is extended in the time domain by a factor corresponding to the reciprocal of the amount of the applied compression using a technique termed globally uniform stretching. Experiments were carried out to verify the proposed method on an ultrasound elasticity phantom with radio-frequency data acquired from a diagnostic ultrasound clinical scanner. It is found that the new method improves the contrast-to-noise ratio by a factor of up to about 1.8 compared to a conventional strain imaging method that employs a reversed gray color map without globally uniform stretching.

• The Journal of the Acoustical Society of Korea
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• v.28 no.6
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• pp.526-533
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• 2009

This paper proposes a formant synthesis method of Haegeum sounds using cepstral envelope for spectral modeling. Spectral modeling synthesis (SMS) is a technique that models time-varying spectra as a combination of sinusoids (the "deterministic" part), and a time-varying filtered noise component (the "stochastic" part). SMS is appropriate for synthesizing sounds of string and wind instruments whose harmonics are evenly distributed over whole frequency band. Formants extracted from cepstral envelope are parameterized for synthesis of sinusoids. A resonator by Impulse Invariant Transform (IIT) is applied to synthesize sinusoids and the results are bandpass filtered to adjust magnitude. The noise is calculated by first generating the sinusoids with formant synthesis, subtracting them from the original sound, and then removing some harmonics remained. Linear interpolation is used to model noise. The synthesized sounds are made by summing sinusoids, which are shown to be similar to the original Haegeum sounds.

• The Journal of the Acoustical Society of Korea
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• v.28 no.3
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• pp.279-284
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• 2009

Speaker verification determines whether the claimed speaker is accepted based on the score of the test utterance. In recent years, methods based on Gaussian mixture models and universal background model have been the dominant approaches for text-independent speaker verification. These speaker verification systems based on these methods provide very good performance under laboratory conditions. However, in real situations, the performance of speaker verification system is degraded dramatically. For overcoming this performance degradation, the feature recombination method was proposed, but this method had a drawback that whole sub-band feature vectors are used to compute the likelihood scores. To deal with this drawback, a modified feature recombination method which can use each sub-band likelihood score independently was proposed in our previous research. In this paper, we propose a sub-band weighting method based on sub-band signal-to-noise ratio which is combined with previously proposed modified feature recombination. This proposed method reduces errors by 28% compared with the conventional feature recombination method.

• Journal of Broadcast Engineering
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• v.12 no.1 s.34
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• pp.28-40
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• 2007

In this paper, we proposes an idea for improved sound characteristic which decreasing a problem in previous reverberation algorism structure. To later reflection sound, proposed new reverberation structure, using a lopped and nested all-pass delay line, and it is designed to improve an natural concert hall sound. In addition, In order to have best imaginary sound effect, we extracted the factors by controlling each delay line's delay time, and we realized a proposed new algorithm by using general-purpose DSP. Through several experimental cases, we observed better effect on improvement of linear flatten and reverberation density and decreasing about colorlessness and non-linear sound at previous proposed model about impulse input.

• The Journal of the Korea Contents Association
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• v.16 no.8
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• pp.202-208
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• 2016

The scope of activities of the visually impaired is increasing. But they are not easy to visit the destination safely because the building was complicated and larger than ever. There is a guide system for visually impaired such as GPS and Audio alerts, Braille guide block, Acoustic signaller, etc. But they are not suitable for indoor because most of them are the outdoor guide system. Therefore, in this paper, I propose a system that provides guidance information to the visually impaired using Voice Technology, TTS (Text to Speech) and Haptic Technology, Beacon based on the wireless sensor networks. It informed the visually impaired of guidance information about destination such as distance, height, and obstacle to the destination using the generalized smart phone. The user could be received guide info about searches for the optimal route to the destination using the TTS technology and Haptic technology in test result of the proposed system.

• Journal of the Korean Society for Nondestructive Testing
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• v.24 no.2
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• pp.136-141
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• 2004

A torsional guided wave mode was applied to detect a crack in a pipe. An array of electromagnetic acoustic transduce. (EMAT that can generate and receive torsional guided ultrasound with the frequency of 200kHz was designed and fabricated for testing a pipe of 2.5 inch diameter Artificial notches with various depths were fabricated in a bent feeder pipe mock-up and the detectability was examined from the distance of 2m of the specimen. The axial notches with the depth of 5% of wall thickness were successfully detected by a torsional mode (T(0,1)) generated by the EMAT However, it was found that the depth of defects was not related to the signal amplitude.

• Journal of Korean Society of Transportation
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• v.27 no.2
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• pp.189-197
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• 2009

The purpose of this study is to analyze the relationships between the traffic accidents of elderly pedestrians and barrier-free facilities in the case of Cheongju. First, many accidents of elderly pedestrians were determined to occur in the road and during crossing. Second, the correlation analysis shows that the paving conditions, guiding blocks and embossed blocks have impacts on elderly safety. Finally, the logistic regression model, which is statistically significant (chi-square =0.000, Nagelkerke =0.198), was developed, and includes the paving conditions, bollards, audible signals and remaining time signs as the independent variables. The variables, with the exception of the existence of bollards, are all analyzed to have positive impacts to elderly safety.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.1
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• pp.86-96
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• 2013

Unified Speech and Audio Coding (USAC) is the speech/audio codec with the best quality, approved on Final Draft International Standard (FDIS) at MPEG meeting in 2011. Since MPEG conventionally standardizes only the decoder, it is not easy to study on the encoder technologies. Furthermore, Reference Model(RM) shows extremely poor performance. To solve these problems, the open source project(JAME) proposes the methods to make the improved performance of main encoder technologies in USAC. Especially, this paper introduces the encoder modules: the signal classifier for selective operation between two coders, the psychoacoustic model in frequency domain, and window transition technology. Finally, the results of verification test for FDIS and the performance of Common Encoder are appended.

• The Journal of the Acoustical Society of Korea
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• v.19 no.3
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• pp.51-56
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• 2000

Is this paper, we studied on the automatic speech control system in real-time windows environment using voice recognition. The applied reference pattern is the variable DMS model which is proposed to fasten execution speed and the one-stage DP algorithm using this model is used for recognition algorithm. The recognition vocabulary set is composed of control command words which are frequently used in windows environment. In this paper, an automatic speech period detection algorithm which is for on-line voice processing in windows environment is implemented. The variable DMS model which applies variable number of section in consideration of duration of the input signal is proposed. Sometimes, unnecessary recognition target word are generated. therefore model is reconstructed in on-line to handle this efficiently. The Perceptual Linear Predictive analysis method which generate feature vector from extracted feature of voice is applied. According to the experiment result, but recognition speech is fastened in the proposed model because of small loud of calculation. The multi-speaker-independent recognition rate and the multi-speaker-dependent recognition rate is 99.08% and 99.39% respectively. In the noisy environment the recognition rate is 96.25%.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.42 no.2
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• pp.432-443
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• 2017

The primary goal of the MPEG-H 3D Audio standard is to provide immersive audio environments for high-resolution broadcasting services such as UHDTV. This standard incorporates a wide range of technologies such as encoding/decoding technology for multi-channel/object/scene-based signal, rendering technology for providing 3D audio in various playback environments, and post-processing technology. The reference software decoder of this standard is a structure combining several modules and can operate in various modes. Each module is composed of independent executable files and executed sequentially, real time decoding is impossible. In this paper, we make DLL library of the core decoder, format converter, object renderer, and binaural renderer of the standard and integrate them to enable frame-based decoding. In addition, by measuring the computation complexity of each mode of the MPEG-H 3D-Audio decoder, this paper also provides a reference for selecting the appropriate decoding mode for various hardware platforms. As a result of the computational complexity measurement, the low complexity profiles included in Korean broadcasting standard has a computation complexity of 2.8 times to 12.4 times that of the QMF synthesis operation in case of rendering as a channel signals, and it has a computation complexity of 4.1 times to 15.3 times of the QMF synthesis operation in case of rendering as a binaural signals.