• The Journal of the Acoustical Society of Korea
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• v.32 no.4
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• pp.329-334
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• 2013

Order spectrum analysis is widely used to grasp the features of noises due to powertrain system including engine and intake/exhaust system. It is known from many previous researches that order components related to the first and second firing frequencies of engine considerably affect the noise of car interior. The purpose of this paper is to find out the difference in sound quality: Pleasantness of car interior noise according to the change of its order spectrum. For this, car interior noises of 6-cylinder and 4-cylinder engines are recorded and their order spectrum levels are modified by applying adaptive digital filters. After subjective listening test employing paired comparison method is conducted, it is investigated that the level change of half-order components is a noticeable factor to improve Pleasantness of the car interior noises whereas level decrease of firing order does not always give the positive effect on its sound quality.

• The Journal of the Acoustical Society of Korea
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• v.24 no.5
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• pp.248-254
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• 2005

In this Paper, the structure of a split bandwidth wideband speech coder and its highband coder for tone qualify elevation are Proposed. The lowband and highband by the split bandwidth method are encoded independently applying the G.729E and MLT (Modulated Lapped Transform) residual model. In the highband structure which is encoded by low bit rate of 4kbps, the MLT residual signals are distinguished to voice and unvoice signal . The voice signals are applied to MLT peak picking method by lowband pitch period. Because transformed MLT residual signals are represented by periodic signal that have periodic peak. The unvoice signals are applied to MLT which linear prediction spectral response is added and do vector quantization. Performance for proposed 15.8kbps wideband speech coder was verified through subjective listening test.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.11 no.4
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• pp.287-295
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• 1986

In this paper, a SBC(sub-band-coding) is proposed to code a speech signal for a digital mobile radio and a robustness of speech quality of the SBC over the Rayleigh fading channel is investigated via a computer simulation. First the Rayleigh fading channel and 16-ary DPSK receiver models are presentes and verified its validitties by comparing with theoretical values. Three different measures: SNR, LPC distance measure and subjective listening test, were used to evaluate the effects due to the Rayleigh fading channel errors. From the results of computer simulation at BER=$10_{-3}$, $10_{-2}$, 5$ imes$$10_{-2}$, it was found that the speech remained quite intelligible at BER=$10_{-2}$and the link is still usuable even at BER=5$ imes$$10_{-2}$ Thus it was concluded that the SBC can be applicable to the digital mobile radio on the Rayleigh fading channel error in the range of $10_{-4}$~$10_{-2}$ without emplowing any error correction codes.

• The Journal of the Acoustical Society of Korea
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• v.35 no.3
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• pp.202-207
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• 2016

Most of noises from cars or home appliances accompany successive impulsive noise due to repeated operation. A human auditory system is able to perceive the successive impulsive noise as either a set of independent noise or amplitude modulated noise according to its occurrence period. This study is to identify main influence factors on understanding of impulsive characteristics and find most appropriate sound quality metrics to express the successive impulsive noise. To do this, the successive impulsive noises were designed and utilized to perform a listening test for identifying conditions where successive impulsive noise can be recognized to have impulsive characteristics. These results were analyzed with sound quality metrics such as loudness, fluctuation strength, and roughness in order to compare the subjective results with the objective results. Consequently, the results revealed that the successive impulsive noise exhibits impulsive characteristics when its occurrence frequency is less than 50 Hz. It was also observed that roughness and fluctuation strength results are not applicable to express the successive impulsive noise because they heavily depend on the amplitude modulation characteristics. On the other hand, loudness results are considered to be useful as an evaluation factor of the successive impulsive noise through the use of loudness limen because it does not depend on the amplitude modulation characteristics.

• The Journal of the Acoustical Society of Korea
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• v.30 no.5
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• pp.281-294
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• 2011

Physics-based sound synthesis usually requires high computational costs and this results in a restriction of its use in real-time applications. This motivates us to implement the sound synthesis algorithm of plucked-string instruments using multi-core processor architectures and determine the optimal processing element (PE) configuration for the target instruments. To determine the optimal PE configuration, we evaluate the impacts of a sample-per-processing element (SPE) ratio that is defined as the amount of sample data directly mapped to each PE on system performance and both area and energy efficiencies using architectural and workload simulations. For the acoustic guitar, the highest area and energy efficiencies are achieved at a SPE ratio of 5,513 and 2,756, respectively, for the synthesis of musical sounds sampled at 44.1 kHz. In the case of the classical guitar, the maximum area and energy efficiencies are achieved at a SPE ratio of 22,050 and 5,513, respectively. In addition, the synthetic sounds were very similar to original sounds in their spectra. Furthermore, we conducted MUSHRA subjective listening test with ten subjects including nine graduate students and one professor from the University of Ulsan, and the evaluation of the synthetic sounds was excellent.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.2C
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• pp.246-253
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• 2010

In this paper, a method of removing the space skew and cross-talk cancellation is proposed where the self-generated signals from the subject are used to obtain the subject's location. In the proposed method, the good spatial sound image is maintained even when the listener moves from the sweet spot. Two major parts of the proposed method are as follows: listener position tracking using the stimuli from the subject and removal of the space skew and cross-talk signals. Listener position tracking is achieved by estimation of the time difference of arrival (TDoA). The position of the listener is then computed using the Talyer-series estimation method. The head-related transfer functions (HRTF) are used to remove the space skew and cross-talk signals, where the direction of the HRTF is given by the one estimated from the listener position tracking. The performance evaluation is carried out on the signals from the 100 subjects that are composed of the 50 female and 50 male subjects. The positioning accuracy is achieved by 70%~90%, under the condition that the mean squared positioning error is less than $0.07m^2$. The subjective listening test is also conducted where the 27 out of the 30 subjects are participated. According to the results, 70% of the subjects indicates that the overall quality of the reproduced sound from the proposed method are improved, regardless of the subject's position.