• Journal of Electrical Engineering and Technology
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• v.10 no.5
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• pp.2126-2134
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• 2015

The present study suggests a localization method for the first (S₁) and the second (S₂) feature of heart sounds, based on an algorithm involving frequency filtering, energy detection, and interval regulation. Localization accuracy was evaluated by comparing the algorithm with the traditional Hilbert transform-based localization method. Results show that the sensitivity and the positive predictivity value of proposed method, respectively, were 97.27 % and 99.94 % in S₁ detection and 94.99 % and 100 % in S₂ detection.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.1231-1234
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• 1998

Many signal processing methods of widening the sound image for spatial impression have been studied. Most typical methods of widening the sound image are related to the phase shifting and precedence effect. However, these methods are not effective in center sound image. As listener's position moves from center to outside, the center sound image is shifted to the speaker. That is to say, the directional localization of center sound image is unstable. In this paper, we propose a television audio system including center speaker, and analyze the role of center speaker using theory of Makida and precedence effect. In experiments, we confirm the usefulness of the center speaker for the stability of center sound image.

• 제어로봇시스템학회:학술대회논문집
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• 2001.10a
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• pp.117.2-117
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• 2001

In this paper, 3D sound rendering using multichannel speakers is studied. Virtual 3D sound technology has mainly been researched with binaural system. The conventional binaural sound systems reproduce the desired sound at two arbitrary points using two speakers in 3DD space. However, it is hard to implement the localization of virtual source at back/front and top/below positions because the HRTF of an individual is unique just like the fingerprint. Most of all, the HRTF is highly sensitive to the elevation change. Multichannel sound systems have mainly been used to reproduce the sound field picked up over a certain volume rather than at specific points. Moreover, multichannel speakers arranged in 3-D space produce a much better performance of ...

• Proceedings of the IEEK Conference
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• 2002.06e
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• pp.113-116
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• 2002

In this paper, we propose the sound-mapping algorithm of the detected obstacle by ultrasonic sensors. We apply this algorithm to a Obstacle alarm for the visually impaired. In our system, we acquire obstacles information using ultrasonic sensors, and transform two-dimensional and distance information into sound-imaging information and vibrator with azimuth (direction) and distance. We implement this system with ultrasonic sensors to more effective expression of the obstacle information. The distance of an obstacle can be expressed by sound pressure level, and azimuth of the obstacles can be expressed by inter-aural time difference (ITD) and inter-aural level difference (ILD) that are two important cues in a binaural system. These are the principal cues for sound localization, to detect sound source. In this system, the obstacle is substituted with a sound source. The visually impaired receive sound information of obstacles by headphone.

• Phonetics and Speech Sciences
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• v.3 no.3
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• pp.109-114
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• 2011

To estimate sound direction of arrival with a pair of microphones, a method based on Time Difference of Arrival (TDOA) estimation using the Cross Power Spectrum Phase (CPSP) function is largely used due to its simplicity and good performance. In this paper, we investigate CPSP maximum values for various SNRs and adverse environments, and propose a novel method to improve the estimation performance of sound direction of arrival. The proposed method applies a threshold to the CPSP values and increases the reliability of the estimated sound direction. Through computer simulation for various SNRs, we validate the effectiveness of the proposed method. When the threshold was set to 0.1, more than 90% of success rate of sound direction of arrival estimation has been achieved for directions of $10^{\circ}$, $40^{\circ}$, $70^{\circ}$ from the source location even with reverberation times of 0.1s.

• Journal of Institute of Control, Robotics and Systems
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• v.22 no.7
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• pp.562-569
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• 2016

This paper proposes a navigation system with a robust localization method for an underwater unmanned vehicle. For robust localization with IMU (Inertial Measurement Unit), a DVL (Doppler Velocity Log), and depth sensors, the EKF (Extended Kalman Filter) has been utilized to fuse multiple nonlinear data. Note that the GPS (Global Positioning System), which can obtain the absolute coordinates of the vehicle, cannot be used in the water. Additionally, the DVL has been used for measuring the relative velocity of the underwater vehicle. The DVL sensor measures the velocity of an object by using Doppler effects, which cause sound frequency changes from the relative velocity between a sound source and an observer. When the vehicle is moving, the motion trajectory to a target position can be recorded by the sensors attached to the vehicle. The performance of the proposed navigation system has been verified through real experiments in which an underwater unmanned vehicle reached a target position by using an IMU as a primary sensor and a DVL as the secondary sensor.

• The Journal of Korea Robotics Society
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• v.3 no.3
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• pp.165-175
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• 2008

For an advanced intelligent service, the need of HRI technology has recently been increasing and the technology has been also improved. However, HRI components have been evaluated under stable and controlled laboratory environments and there are no evaluation results of performance in real environments. Therefore, robot service providers and users have not been getting sufficient information on the level of current HRI technology. In this paper, we provide the evaluation results of the performance of the HRI components on the robot platforms providing actual services in pilot service sites. For the evaluation, we select face detection component, speaker gender classification component and sound localization component as representative HRI components closing to the commercialization. The goal of this paper is to provide valuable information and reference performance on appling the HRI components to real robot environments.

• Proceedings of the IEEK Conference
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• 2000.06d
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• pp.102-105
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• 2000

This paper propose new sound localization algorithm that calculates TDOA(Time Difference Of Arrival) between the two received signals via two microphone array, The proposed Subband CPSP is a development of Previous CPSP method using subband approach. It first split the received microphone signals into three frequency bands and then calculates subband CPSP with corresponding SNR weights. This type of algorithm, Subband CPSP, can provide more accurate TDOA estimation results because it limits the effects of environmental noise within each subband. To verify the performance of the proposed Subband CPSP algorithm, computer simulation was conducted and it was compared with previous CPSP method. From the both simulation results, the proposed Subband CPSP is superior to previous CPSP algorithm more than accuracy for TDOA estimation.

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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• pp.215-220
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• 2005

Nowadays, various speech processing systems are being introduced in the fields of robotics. However, real-time processing and high performances are required to properly implement speech processing system for the autonomous robots. Achieving these goals requires advanced hardware techniques including intelligent software algorithms. For example, we need nonlinear amplifier boards which are able to adjust the compression radio (CR) via computer programming. And the necessity for noise reduction, double-buffering on EPLD (Erasable programmable logic device), simultaneous multi-channel AD conversion, distant sound localization will be explained in this paper. These ideas can be used to improve distant and omni-directional speech recognition. This speech processing system, based on embedded Linux system, is supposed to be mounted on the new home service robot, which is being developed at KIST (Korea Institute of Science and Technology)

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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• pp.872-876
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• 2005

We propose a real-time post processing of nonlinear amplified signal to improve voice recognition in remote talk. In the previous research, we have found the nonlinear amplification has unique advantage for both the voice activity detection and the sound localization in remote talk. However, the original signal becomes distorted due to its nonlinear amplification and, as a result, the rest of sequence such as speech recognition show less satisfactorily results. To remedy this problem, we implement a linearization algorithm to recover the voice signal's linear characteristics after the localization has been done.