• Journal of KSNVE
• /
• v.8 no.5
• /
• pp.879-886
• /
• 1998

An automated calibration system of sound level meters was developed and tested. As a standard sound source, the speaker unit(Forstex FE208) cabineted by 440$\times$390$\times$490 $\textrm{mm}^3$(LHW) volume wood box was adopted. Including this source, the driving part was found out to have a good linearity of sound pressure output vs AC voltage input. The Hybrid-Bisect/Newton-Raphson method modified by the linearity was adopted as a searching algorithm. Uisng GPIB interface, the console PC make the control, measurements, and calculations and finally make the accumulation of useful data and results automatically by the instructon in the program coded by C languate. Several trials of automatic calibration using this developed system give the reliable results.

• Transactions of the Korean Society for Noise and Vibration Engineering
• /
• v.15 no.9 s.102
• /
• pp.1003-1008
• /
• 2005

The acoustic absorption of a multiple layer perforated plate system is very good near the resonance frequency region, while it has been regarded as a demerit that its frequency bandwidth is considerably narrow. In order to overcome such a demerit, the parallel perforated plates with different porosities are proposed. The sound absorption of such system composed of a parallel perforated plate is calculated by an equivalent electroatoustic circuit approach and validated by comparing the calculated absorption coefficients with those measured by the two-microphone impedance tube method. The sound absorptive characteristics and performance of parallel perforated plate systems are discussed from a standpoint of frequency bandwidth related with sound absorption.

• Transactions of the Korean Society for Noise and Vibration Engineering
• /
• v.24 no.2
• /
• pp.108-116
• /
• 2014

With the increasing the importance of emotional quality of vehicle, the sound quality of systems with electric motor components has become increasingly important. Electric motors are used for windows, seats, sun roof, mirrors, steering columns, windshield wiper and climate control blowers, etc. In this paper, a study was conducted to identify sound quality factors that contribute to customer's satisfaction and preference of the window lift system. Jury test for subjective evaluation was carried out and sound quality index was developed. Averaged sound pressure level and sharpness were significant factors when glass moves down. Also, maximum loudness at stop section and averaged loudness were significant factor when glass moves up. Next, noise source identification was carried out using beam forming method during glass transferred section and impulsive noise at stop section. Several improvement methods were applied using the source identification result. And finally, the degree of sound quality improvement was judged using sound quality index.

• The Transactions of the Korea Information Processing Society
• /
• v.5 no.3
• /
• pp.851-861
• /
• 1998

Korean language is to be characterized by the rich vocal sound variation. In order to increase the probability of vocal sound recognition and to provide a natural vocal sound synthesis, a systematic and thorough research into the characteristics of Korean language including its vocal sound changing rules is required. This paper addresses an effective way of vocal sound recognition and synthesis by providing the design and implementation of the Korean vocal sound variation rule. The regulation we followed for the design of the vocal sound variation rule is the Phonetic Standard(Section 30. Chapter 7) of the Korean Orthographic Standards. We have first factor out rules for each regulations, then grouped them into 27 groups for eaeh final-consonant. The Phonological Change Processing System suggested in the paper provides a fast processing ability for vocal sound variation by a single application of the rule. The contents of the process for information augmented to words or the stem of innected words are included in the rules. We believe that the Phonological Change Processing System will facilitate the vocal sound recognition and synthesis by the sentence. Also, this system may be referred as an example for similar research areas.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
• /
• 2005.11b
• /
• pp.186-189
• /
• 2005

Since human listening is very sensitive to sound, a subjective index of a sound quality is required. Therefore, in the analysis for each situation, the sound evaluation is composed with sound quality factor. Many researchers spends their effort to make a more reliable and more accurate of sound in term of sound quality index for various system noise. In this study a reliable index is constructed and analyzed using correlation analysis, regression analysis and weighting factor for each sound quality factor. We have made the sound quality index that agrees with more than human subjective sensitivity which apply to various sound quality metrics. Also we applied a 'grade' metric to jury for sound evaluation, analyzed relation between sound duality index and sound quality grade. Then we will judge the sound quality level according to the sound quality grade scheme.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
• /
• 1995.10a
• /
• pp.7-15
• /
• 1995

Sound pressure and particle velocity are the most essential quantities prescribing a sound field; they correspond to voltage and electric current respectively, in electric system. As electric power is the product of voltage and electric current, sound intensity is the product of sound pressure and particle velocity and it means the acoustic power passing through a unit area in a sound field. Although the definition of sound intensity is very simple as mentioned above, the method of measuring this quantity has not been realized for a long time, because it has been very difficult to measure the particle velocity simultaneously with the sound pressure. Owing to the recent development of such technologies as transducer production and digital signal processing, it has finally been realized. According to the sound intensity(SI) method, the sound power flow in an arbitrary sound field can be directly measured as a vector quantify. In this paper, the principle of the SI method is briefly explained at first and some examples of its application made in the author's laboratory are introduced.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
• /
• 2007.05a
• /
• pp.515-518
• /
• 2007

SAFRS(Spontaneous Acoustic Field Reproduction System) is a system to sense changes of surroundings and produce sounds which can go well with environment elements sensed by the system in to the space. The sounds were judged by individual evaluation and, the classification of the preferred sounds according to the mood of the space was suggested in the former study. Effectiveness of SAFRS with field application was validated by prior studies which dealt with researching acoustic environment, evaluating images of sounds, and rating environment with existence and nonexistence of sound resources such as fountains and the system after applied in D university. In this study, for more effective field application of SAFRS, research for the acoustic environment around sound resources and subjective evaluation of the preference of the sounds from the resources were made and it was considered that the results of the experiments should be primary information to propose proper sound level to be offered by the system. The results of the study are as follows; 1) It was considered that the ambience of the center road was dependent upon produced sounds by the system and water sounds of the fountain and that of walk way was mostly dependent upon produced sounds. 2) The results of the subjective evaluation showed that the distance from sound resources was suggestive; the more distant from produced sounds the less full and clear the sounds, the less distant from the sounds of water the more delight and idyllic ambience, and the less distant from the forest the more idyllic ambient and diversity. 3) The results upwards were telling that an average value of six elements for the evaluation was even at the place set back 10.2m from center road and walk way. And harmony of all sounds of the place should be considered to propose suitable sound level of SAFRS.

• Transactions of the Korean Society for Noise and Vibration Engineering
• /
• v.17 no.6 s.123
• /
• pp.547-552
• /
• 2007

SAFRS(spontaneous acoustic field reproduction system) is a system to sense changes of surroundings and produce sounds which can go well with environment elements sensed by the system in to the space. The sounds were judged by individual evaluation and, the classification of the preferred sounds according to the mood of the space was suggested in the former study. Effectiveness of SAFRS with field application was validated by prior studies which dealt with researching acoustic environment, evaluating images of sounds, and rating environment with existence and nonexistence of sound resources such as fountains and the system after applied in D university. In this study, for more effective field application of SAFRS, research for the acoustic environment around sound resources and subjective evaluation of the preference of the sounds from the resources were made and it was considered that the results of the experiments should be primary information to propose proper sound level to be offered by the system. The results of the study are as follows; 1) It was considered that the ambience of the center road was dependent upon produced sounds by the system and water sounds of the fountain and that of walk way was mostly dependent upon produced sounds. 2) The results of the subjective evaluation showed that the distance from sound resources was suggestive; the more distant from produced sounds the less full and clear the sounds, the less distant from the sounds of water the more delight and idyllic ambience, and the less distant from the forest the more idyllic ambient and diversity. 3) The results upwards were telling that an average value of six elements for the evaluation was even at the place set back 10.2m from center road and walk way. And harmony of all sounds of the place should be considered to propose suitable sound level of SAFRS.

• Journal of the Korean Society of Fisheries and Ocean Technology
• /
• v.44 no.1
• /
• pp.31-36
• /
• 2008

The underwater sound transmission system(USTS) was experimentally made to monitor the cetacean's appearance by telemetry, and then its system was tested to evaluate its performance from July to October, 2007 at the Kimnyeong berth and the dolphin's breeding ground of Pacific Land in Jeju island, respectively. The results showed that the sweep sound in the trial experiment and the whistle sound of bottlenose dolphin(Tursiops truncatus) were favorably received by telemetry. Therefore, we could confirm the USTS is able to monitor the cetacean's appearance in real time without direct observation at sea within effective range of code division multiple access(CDMA) communication method.

• The Journal of the Acoustical Society of Korea
• /
• v.20 no.4E
• /
• pp.52-60
• /
• 2001

This paper describes real time implementation of subband sound localization system on a floating-point DSP TI TMS320C31. The system determines two dimensional location of an active speaker in a closed room environment with real noise presents. The system consists of an two microphone array connected to TI DSP hosted by PC. The implemented sound localization algorithm is Subband CPSP which is an improved version of traditional CPSP (Cross-Power Spectrum Phase) method. The algorithm first split the input speech signal into arbitrary number of subband using subband filter banks and calculate the CPSP in each subband. It then averages out the CPSP results on each subband and compute a source location estimate. The proposed algorithm has an advantage over CPSP such that it minimize the overall estimation error in source location by limiting the specific band dominant noise to that subband. As a result, it makes possible to set up a robust real time sound localization system. For real time simulation, the input speech is captured using two microphone and digitized by the DSP at sampling rate 8192 hz, 16 bit/sample. The source location is then estimated at once per second to satisfy real-time computational constraints. The performance of the proposed system is confirmed by several real time simulation of the speech at a distance of 1m, 2m, 3m with various speech source locations and it shows over 5% accuracy improvement for the source location estimation.