• Journal of the Korea Institute of Information and Communication Engineering
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• v.21 no.4
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• pp.708-717
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• 2017

Tracking of sound-location is widely used in various area such as intelligent CCTV, video conference and voice commander. In this paper we introduce the real-time sound-location tracking method for smart lecture system using TDoA(Time Difference of Arrival) with orthogonal microphone array on the ceiling. Through discussion on some models of TDoA detection, cross correlation method using linear microphone array is proposed. Orthogonal array with 5 microphone could detect omni direction of sound-location. For real-time detection we adopt the threshold of received energy for eliminating no-voice interval, signed cross correlation for reducing computational complexity. The detected azimuth angles are processed using median filter for lowering the angle deviation. The proposed system is implemented with high performance MCU of TMS320F379D and MEMs microphone module and shows the accuracy of 0.5 and 6.5 in degree for white noise and lectured voice, respectively.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.25 no.9
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• pp.628-636
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• 2015

In this study, theoretical and experimental analyses were performed for creating and moving the zone of quiet(ZoQ) to the ear location of a sitter by using active sound control technique. As the ZoQ is actively created at the location of the error microphone basically with an active sound control system using an algorithm such as the filtered-x least mean square(FxLMS), the virtual microphone control(VMC) method was considered to move the location of the ZoQ to around the sitter`s ear. A chair system with microphones and loudspeakers on both sides was manufactured for the experiment and thus an active headrest against the swept narrowband noise as the primary noise was implemented with a real-time controller in which the VMC algorithm was embedded. After the control experiment with and without the VMC method, the location variation of the ZoQ by analyzing the error signals measured by the error and the virtual microphones. Therefore, it is observed that the FxLMS with the VMC technique can provide the re-location of the ZoQ from the error microphone location to the virtual microphone location. Also it is found that the amount of the attenuation difference between the two locations was small.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2011.04a
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• pp.816-820
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• 2011

A method for increasing the difference of side-lobe level in spherical microphone array is presented. In array signal processing, it is known that narrow interval between sensors can increase the difference between main lobe and side-lobe of array response which eventually increase the source recognition capability. Recent commercial array being used, however, have shown certain limitation in using the number of sensors due to its costs and geometrical size of array. To overcome this problem, we have adapted MEMS sensors into spherical microphone array. To check out the improvement, two different types of spherical microphone array were designed. One array is composed with 32 regular instrument microphones and the other one is 85 MEMS sensors. Simulation and experiments were conducted on a sinusoidal noise source with two arrays. The time history data were analyzed with spherical harmonic decomposition and beamforming technique. 85 MEMS sensors array showed the improved side-lobe level suppression by more than 4 dB above the frequency content of 2 kHz compared to 32-sensor array.

• Journal of Biomedical Engineering Research
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• v.32 no.3
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• pp.198-206
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• 2011

Hearing devices like cochlear implant, vibrant soundbridge, etc. try to offer better sound for people. In hearing devices, several beamformers including conventional directional microphone are applicable to noise reduction. Each beamformer has different directional response and it could change sound intelligibility or quality for listeners. Therefore, we investigated the performance of three beamformers, which are first and second order directional microphone, and broadband beamformer(BBF) with a computer simulation assuming hearing device microphone configuration. We also calculated objective measurements which have been used to evaluate speech enhancement algorithms. In the simulation, a single speech and a single babble noisewere propagated from the front and $135^{\circ}$ azimuth degrees respectively. Microphones were configured in an end-fire array and the spacing was varied in comparison. With 3 cm spacing, BBF had about 3 dB higher enhanced SNR than that of directional microphones. However, enhancement of segmental SNR and frequency weighted segmental SNR were similar between the first order directional microphone and broadband beamformer. In addition when steady state noise was used, broadband beamformer showed the increased performance and had the highest enhanced SNR, and segmental SNR.

• The Journal of the Acoustical Society of Korea
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• v.16 no.1
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• pp.47-53
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• 1997

This paper concerns an efficient approach to forming a beam pattern of a microphone array to deal with broadband signals such as speech in a teleconference. A numerical method is proposed to find updated location of sidelobes for equalizaing the sidelobes via perturbation of array parameters such as array weight or microphone spacing. Thus the microphone array is optimized in a Dolph-Chebyshev sense such that directional or background noises incident in an array visual range are eliminated efficiently. It is shown that perturbation of microphone spacing yields an optimum pattern more appropriate for dealing with broadband signals than that of array weight. Also, a novel method is proposed to find a beam pattern which is robust with respect to sidelobe in a scanning situation. Computer simulation results are presented.

• Korean Journal of Air-Conditioning and Refrigeration Engineering
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• v.14 no.1
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• pp.63-72
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• 2002

The main objective of this study is to propose a practical two-microphone impedance tube method to measure the sound transmission loss for flexible sound isolation sheets without the use of the time-consuming and expensive reverberation room. This method was based on the sound decomposition theory developed by Seybert using the spectral density functions of the incident and reflected sound waves. In order to verify the validity of the experimental results, the measured sound transmission losses from the proposed method were compared with the measured data from the reverberation room method and the calculated data from the theory satisfying the mass law of sound isolation material. The resulted trends of the sound transmission losses versus frequencies for several different sound isolation sheets were almost same for each other and agreed quite well in both methods except at some low frequency region. From the experimental results, it was found that the accuracy of sound isolation capability obtained by two-microphone impedance tube method depends upon the microphone spacing, the distance from the first microphone to the test sample surface and the test sample location.

• Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
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• v.18 no.1
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• pp.26-32
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• 2007

Objective and Background: Laryngeal stroboscope is an useful equipment in evaluation of vocal cord vibration and in early detection of mucosal lesion including invasive cancer of the vocal cord. Recently Lee et al. (2006) developed portable stroboscope using voice as synchronization signal. It has been frequently impaired ability to synchronize the flashes even in normal female. Authors tried to investigate various methods including vibration pick-up, microphone, laryngeal microphone, and contact microphone for development of simple and accurate method like electroglottograph signal. The purpose of this study was to estimate wheher the vibration pick-up is available and is consistent with the signal of EGG. Subjects and Methods: Authors compared the signals between EGG and noncontact method such as voice, contact methods including vibration pick-up, laryngeal microphone, and contact microphone in normal twenty adults (male 10 and female 10). The number of peak in one cycle was compared with the number of the peak in EGG, and the percent of phase difference in the peak was compared with EGG Also, authors tried to investigate which site of vibration pick-up was most effective for synchronization of stobo flashes. Three site including anterior neck below the cricoid cartilage, thyroid ala, and suprahyoid region were analysed. Results: Among various methods for synchronization of strobo flashes, vibration pick-up was most effective method in peak detection. And anterior neck below cricoid cartilage was the most available site of the vibration pick-up. Conclusion: Authors suggest that vibration pick-up is most available and effective method for synchronization of strobo flashes.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.439-442
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• 2000

When we measure the acoustical properties of a room using multiple microphone system, it is important to grasp exact time delay of the early reflections from impulse response pair. But it is often very difficult to identify the early reflections in natural shape, because a waveform may be deformed due to the characteristics of a sound source loudspeaker, microphone and reflected wall and overlapping of plural waveform. In this paper to obtain more accurate and enough early reflections, we propose the brand-new five-channel sound receiving system and introduce peak-detecting algorithm. The system has microphones mounted at the origin and four points of a regular tetrahedron. The newly introduced peak-detecting algorithm can show exact peak position in each channel, in spite of deformation due to reflected walls, loudspeaker and microphone.

• Proceedings of the IEEK Conference
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• 2001.09a
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• pp.503-506
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• 2001

In this paper, we describe the implementation of a microphone input level control algorithm and the speech improvement with this level controller in personal computer environment. The volume of speech obtained through a microphone affects the speech recognition rate directly. Therefore, proper input volume level control is desired fur better recognition. We considered some conditions for the successful volume controller implementation firstly, then checked its usefulness on our speech recognition system with common office environment speech database. Cepstral mean subtraction is also utilized far the channel-effect compensation of the database. Our implemented controller achieved approximately 50% reduction, i.e., improvement in speech recognition error rate.

• Proceedings of the KSPS conference
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• 2002.11a
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• pp.187-190
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• 2002

In this paper, a speech enhancement system using microphone array with MMSE-STSA (Minimum Mean Square Error-Short Time Spectral Amplitude) estimator based post-processing is proposed. Speech enhancement is first carried out by conventional delay-and-sum beamforming (DSB). A new MMSE-STSA estimator is then obtained by refining MMSE-STSA estimators from each microphone, which is applied to the output of conventional DSB to obtain additional speech enhancement. Computer simulation for white and pink noises show that the proposed system is superior to other approaches.