• The Journal of the Acoustical Society of Korea
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• v.37 no.5
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• pp.343-348
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• 2018

This paper proposes a simple procedure for estimating the acoustical parameter values in an occupied classroom from measurements in the unoccupied classroom. The total sound absorption in an unoccupied classroom can be calculated from measured reverberation times in the room. The expected occupied absorption can be calculated using equation that was obtained in a previous study (Choi, 2016) by fitting a linear regression line to a plot of total occupied absorption versus the corresponding unoccupied total absorption values measured in 12 university classrooms. The ratios of occupied-to-unoccupied sound absorption are used to predict increments in the values of acoustical parameters when occupants are added to the rooms. Occupied values of acoustical parameters can be estimated from unoccupied values and the change in total room absorption due to adding occupants.

• Journal of Broadcast Engineering
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• v.18 no.6
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• pp.919-926
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• 2013

The acoustic signal from a distance sound source in an enclosed space often produces reverberant sound that varies depending on room impulse response. The estimation of the level of reverberation or the quality of the observed signal is important because it provides valuable information on the condition of system operating environment. It is also useful for designing a dereverberation system. This paper proposes a speech quality estimation method based on the harmonicity of received signal, a unique characteristic of voiced speech. At first, we show that the harmonic signal modeling to a reverberant signal is reasonable. Then, the ratio between the harmonically modeled signal and the estimated non-harmonic signal is used as a measure of standard room acoustical parameter, which is related to speech clarity. Experimental results show that the proposed method successfully estimates speech quality when the reverberation time varies from 0.2s to 1.0s. Finally, we confirm the superiority of the proposed method in both background noise and reverberant environments.

• KIEAE Journal
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• v.7 no.3
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• pp.63-70
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• 2007

It is known that early reflections, their energy and delay times after the arrival of direct sound are important factors for speech intelligibility. In this basis, acoustical parameters like D50 and C80 had been proposed and are widely used for assessing the listening condition of rooms. These parameters are focused on the fraction of the early energy to the total, regardless of the spatial characteristics of the early reflections. This means that all the early reflections, arrived in certain time boundary. from front, behind, down and upside have the same impact on speech intelligibility. From the questionable simplicity, the influence of the direction of early reflections on speech intelligibility is examined in this study. A computer simulation speech intelligibility test, conducted for 22 university students, found that the reflection of vertical direction with method of the Paired comparison also the preference of 0.746 degree was visible an increase.

• The Journal of the Acoustical Society of Korea
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• v.31 no.3
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• pp.115-128
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• 2012

This paper investigates the effects of diffuser on the acoustical parameters in music hall with consideration of the result of scattering coefficient measurement. A scale model hall of 600 seats with orchestra shell was used for experiments. The materials of 1:50 scale model was chosen through absorption coefficient measurement based on ISO 354. The model was matched to the computer simulation model in terms of reverberation time. In order to evaluate the effect of diffuser location, the measurements were accomplished with and without diffusers according to 7 configurations by diffuser-installed region; sidewall, balcony front, ceiling and so on. The following acoustical parameters were extracted from each measurement case; Reverberation time (RT), Early decay time (EDT), Clarity (C80), Center time (Ts), Sound strength (G) and Temporal diffusion (TD) from the auto-correlation function (ACF) of impulse responses. As a result, the absorption power and diffusion power were increased with number of diffusers. Accordingly RT, EDT and G were decreased by diffuser and the redirection of reflections was occurred briskly. Averaged TD was 6.05 to 6.30 by measurement cases. RT was found to be the most related factor to diffusion power (R = 0.94). The correlation between TD and EDT was high (R = 0.73). In addition, the effects of diffuser-installed location were discussed in terms of acoustical parameter variation.

• The Journal of the Acoustical Society of Korea
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• v.24 no.2
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• pp.103-110
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• 2005

Providing adequate amount of spatial impression for spaciousness) has been known to be one of the most important design considerations for the good acoustics of rooms for music. and the measurement, of room acoustics using parameters. such as LEF and IACC, forms an essential part of such evaluation. However. it is unavoidable to use different transducers (figure of eight microphones. head and torso) for the measurement of each parameter and it tends to make the measurement procedure complicated. The Present work tried to provide a simpler way to measure these binaural room acoustic parameters including monaural ones with a single measurement system using both spatial information collected through a 5-channel microphone and a trained neural network. A computer simulation program, CATT-Acoustic V7.2. which allowed us to obtain exactly the same spatial information as a 5-channel microphone was used. since it requires quite a large amount of data for practical training of a neural network. Since each reflection has different energy. delay and direction, energy should be integrated properly. the concept of ray tracing method was applied inversely in this work. Also applying weightings according to the delay times was considered in this work. Finally, predicted results were compared with the measured data md their correlations were analyzed and discussed.

• 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 Korean Society for Noise and Vibration Engineering Conference
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• 2004.11a
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• pp.1028-1031
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• 2004

Last target of architectural acoustics is that people wish to convey voice effectively from the space adaptively in use purpose in building. But, to how exactly through space sound source that wish to deliver from indoor can be passed does quantification sound estimation method is proposing various kinds physical parameter to estimate degree of voice definition (Speech articulation) and reverberation. Result that evaluate sound source about MLS signal and Impluse signal by pistol in this treatise could know that converge in MLS and measurement error extent about reverberation time(RT) of sound benevolent person. And value is thought there is problem showing change irregularly about sound benevolent person of D50, C80 etc. Finally, in case is spread sound field in difference of sound pressure level, when measure about change of sound pressure level, sound benevolent person could know that there is no different effect.