• ETRI Journal
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• v.44 no.1
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• pp.125-134
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• 2022

Visible light communication (VLC) systems incorporate ambient lighting and wireless data transmission, and the experienced channel in indoor VLC is a major topic that should be examined for reliable communication. In this study, it is realized that multiple transmitters in classical alignment are the forceful factors for channel characteristics. In the frequency band, fluctuations with sudden drops are observed, where the fluctuation shape is related to the source layout and receiver location. These varying frequency-selective channels need solutions, especially for mobile users, because sustained channel estimation and equalization are necessary as the receiver changes its location. It is proven that using light-emitting diodes (LEDs) with highly directional beams as sources or using a detector with a narrow field of view (FOV) in the receiver may help partially alleviate the problem; the frequency selectivity of the channel reduces in some regions of the room. For flat fading channel characteristics all over the room, LEDs should be aligned in hexagonal cellular structure, and detector FOV should be arranged according to the cell dimension outcomes.

• 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.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.39-40
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• 2006

This paper modeled UWB system transmission channel in indoor environment. For channel modeling, we extracted transmission channel parameter, and then measured from 10MHz to 16.01GHz by using method of CIR(Channel Impulse Response) and CTF(Channel Transfer Function) in small room($5.5{\times}5.5m^2$) environment.

• Journal of Korean Society for Atmospheric Environment
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• v.23 no.E2
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• pp.66-73
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• 2007

The introduction of tracer gas techniques to ventilation studies in indoor environments provides valuable information that used to be unattainable from conventional testing environments. Data acquisition systems (DASs) containing analogue-to-digital (A/D) converters are usually used to function the key role that records signals to storage in digital format. In the testing process, there exist a number of components in the measuring equipment which may produce system-based inference to the monitored results. These unwanted fluctuations may cause significant error in data analysis, especially when non-linear algorithms are involved. In this study, a pre-processor is developed and applied to separate the unwanted fluctuations (noise or interference) in raw measurements and to reduce the uncertainty in the measurement. Moving average, notch filter, FIR (Finite Impulse Response) filters, and IIR (Infinite Impulse Response) filters are designed and applied to collect the desired information from the raw measurements. Tracer gas concentrations are monitored during leakage and ventilation tests in the model test room. The signal analysis functions are introduced to carry out the digital signal processing (DSP) work. Overall the FIR filters process the $CO_2$ measurement properly for ventilation rate and mean age of air calculations. It is found that, the Kaiser filter was the most applicable digital filter for pre-processing the tracer gas measurements. Although the IIR filters help to reduce the random noise in the data, they cause considerable changes to the filtered data, which is not desirable.

• Journal of Korean Association for Spatial Structures
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• v.8 no.3
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• pp.65-73
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• 2008

The present work deals with subjective assessments to propose the objective parameter related to the subjective attribute of reverberation in the coupled room varied the aperture opening size. A 1/10 scale model was built and used for the measurements and subjective assessments. For the subjective tests, binaural impulse responses(BIRs) were measured using a dummy head and the measured BIRs were convolved with anechoic recorded music for the presentation over the headphones. The results showed that the perceived reverberation was the largest in the aperture opening size 12.5%(1.97% of mail) room surface area)and decreased with increasing the aperture opening size over 25% (3.94% of main room surface area) in the main room coupled with the secondary reverberant room. In the main room coupled with the secondary absorptive room, there was no changes in the perceived reverberation up to the aperture opening size 6.25%(0.99% of main room surface area) and it gradually decreased with increasing the aperture opening site over 12.5%. The objective parameter, T30/T15, showed a very low correlation with the perceived reverberation in the coupled room varied the aperture opening size but showed a high correlation with a new objective parameter, T30/Tbp, proposed by the authors. Because the late part decay energy more rapidly increases with increasing the aperture opening site than the early and middle part decay energy, the LDT/EDT or T3/T1 is a better quantifier to measure double slopes in the coupled room than the T30/T15.

• The Journal of the Acoustical Society of Korea
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• v.21 no.1
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• pp.42-46
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• 2002

In this paper, a new pre-processing algorithm is proposed to improve the performance of stereo acoustic echo canceller. The proposed algorithm has the improved performance by the estimation error reduction of filter coefficient using input signal which was reduced reverberation of room in the basis MINT (Mu1tip1e-input/output Inverse Theorem) filtering. For real stereo speech signal and real room impulse response the results of simulation, we showed that the proposed method could improved 3∼5 dB ERLE (Echo Return Loss Enhancement) regardless of NLMS (Normalized Least Mean Square) and Projection adaptive algorithm.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2003.11a
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• pp.857-860
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• 2003

We introduced the basic direction of revision for KS F 2805 (method for measurement of sound absorption coefficient in a reverberation room). It is fundamentally equal to ISO 354 (measurement of sound absorption in a reverberation room) which was state of ISO/FDIS 354. Two main points were strongly reviewed. First, the sound absorption coefficient values for Type J test specimen mounting was investigated. Next, the reverberation time difference between the conventional method and new additional one, impulse response method were compared.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.16 no.1 s.106
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• pp.34-39
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• 2006

For the evaluation of the normalized impact sound pressure level, the reverberation time of the receiving room should be measured. This paper deals with the effect of the time constant of FFT analyzer and the measuring points on reverberation time. It is found that the time constant should be in the range between 10 ms and 35 ms. While the effect of measuring points on the reverberation time is significant when the bandwidth is narrow it is negligible in the evaluation of the normalized impact sound pressure level.

• Proceedings of the IEEK Conference
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• 2001.06e
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• pp.189-192
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• 2001

A new configuration of acoustic echo canceler with stepsize predictor and comparator is proposed in this paper. Conventional acoustic echo cancelers using ES(Exponential Step)algorithm has fast convergence speed, but very weak in interference of environment. The proposed stepsize predictor and comparator improve conventional acoustic echo canceler's defects. The Stepsize predictor generates a stepsize value using residual power of error signal. The stepsize comparator selects the stepsize value that is better performance in a acoustic echo canceler using a stepsize decision factor. The Simulation results show superiority of the proposed acoustic echo canceler in environment interference.

• The Journal of the Acoustical Society of Korea
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• v.37 no.5
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• pp.349-355
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• 2018

A large number of parameters have been defined in order to describe and to evaluate the acoustical propeTies in rooms. Reverberation time is an impoTant characteristic of conceT halls, theaters and studios, etc. Over the years, a number of different methods for measuring the reverberation time have been developed, the most common being: the interrupted noise method, the integrated impulse response method, and the method of recording the room response to an impulsive source. However, the reverberation time can be changed by the measurement method, sound source and microphone. Therefore, it is difficult to accurately measure the reverberation time in a room. In this paper, it will be analyzed the interpretation method of the reverberation time and discussed the limitations.