• 한국정밀공학회지
• /
• 제22권8호
• /
• pp.143-150
• /
• 2005

To verify applicability of multi-dimensional spectral analysis (MDSA) fur noise source identification two different approaches which are frequency response and coherent function have been investigated. The coherence function approach appears able to separate the correlated system when the noise sources were coherent. In this study, we identify contribution of structure-borne-noise of vehicle HVAC system using MDSA method. Firstly, to identify the applicability of MDSA method, 4-inputs of vehicle HVAC system were the signals measured by accelerometers attached on the selected noise sources which were composed of blower, evaporator, heater and duct. While 1-output which was driver's position sound was the SPL signals measured by a remote microphone, when the blower motor was operating. We identify efficiency of systems modeled with four Inputs/single output through ordinary coherence function (OCF) and partial coherence function (PCF). As a result of experiment, the blower accounted for $62-88\%$ of the overall level of sound energy density. Also, according to the analysis of acoustic signal and vibration signals measurement, an investigation of the noise source identification in the vehicle HVAC is presented. With the sound intensity method, the major sources of the vehicle HVAC radiation are verified. Also the method of improving the noise reduction is proposed by attaching damping patch access to blower motor and noise reduction is verified.

• 전기전자학회논문지
• /
• 제23권4호
• /
• pp.1360-1365
• /
• 2019

본 논문은 모바일 로봇과 자동제어 시스템에 적용될 수 있는 음원 위치 추적 시스템(Sound Source Localization, SSL)을 보여준다. 대부분 SSL의 기법은 음원 도달 시간차(Interaural Time Difference, ITD)와 음압 레벨의 차이(Interaural Level Difference, ILD)를 구하고, 마이크로폰 배열의 기하학적 원리를 이용하여 위치를 찾게 된다. 하지만 본 논문에서는 음원의 수평 각도를 구하기 위해 깊은 인공 신경망을 기반으로 한 다른 접근법은 제안한다. 인간의 귀를 모방한 로봇의 양쪽 마이크로폰에서 음원의 신호를 채집하여 연구에 사용했다. Network를 학습시키기 위해 양쪽 마이크로폰에서 얻어진 음원의 스펙트럼 분포 차이를 이용하였다. 각 10도 마다 채집한 데이터로 네트워크를 학습시켰고 임의의 각도에서 얻어진 데이터로 결과를 확인했다. 실험 결과 제안한 SSL의 접근 방식은 상당히 가능성이 있는 결과를 보여주었다.

• 대한의용생체공학회:의공학회지
• /
• 제28권6호
• /
• pp.732-743
• /
• 2007

Background noise is one of the major complaints of not only hearing impaired persons but also normal listeners. This paper describes the results of two experiments in which speech recognition performance was determined for listeners with normal hearing and sensorineural hearing loss in noise environment. First, we compared speech enhancement algorithms by evaluation speech recognition ability in various speech-to-noise ratios and types of noise. Next, speech enhancement algorithms by reducing background noise were presented and evaluated to improve speech intelligibility for sensorineural hearing impairment listeners. We tested three noise reduction methods using single-microphone, such as spectrum subtraction and companding, Wiener filter method, and maximum likelihood envelop estimation. Their responses in background noise were investigated and compared with those by the speech enhancement algorithm that presented in this paper. The methods improved speech recognition test score for the sensorineural hearing impaired listeners, but not for normal listeners. The results suggest the speech enhancement algorithm with the loudness compression can improve speech intelligibility for listeners with sensorineural hearing loss.

• 한국소음진동공학회논문집
• /
• 제24권4호
• /
• pp.275-281
• /
• 2014

This study has been analyzed the sound performance of the noise barriers with respect to the configuration of the front perforated panel geometries and the filled absorption materials. Noise barriers' acoustic performance should be required to meet 0.7 of NRC value at least. The global absorbing performance of the barriers have been investigated by changing the opening ratio of the front perforated panel and the absorbing characteristics of the absorbing material using two microphone method. Therefore, It it possible to obtain to increase acoustic performance of the specific frequency ranges by designing the perforate rates of the front panel and absorbing characteristics of the absorbing materials inside, as well. This study try to find out the possibilities of applying the absorbing noise barrier to railway usage.

• 한국소음진동공학회:학술대회논문집
• /
• 한국소음진동공학회 2009년도 춘계학술대회 논문집
• /
• pp.490-495
• /
• 2009

An approach to 3D visualization of multiple sound sources has been developed with the application of a moving array technique. Frequency-domain beamforming algorithm is used to generate a beam power map and the sound source is modeled as a point source. When a conventional delay and sum beamformer is used, it is considered that 2D distribution of sensors leads to have deficiency in spatial resolution along a measurement distance. The goal of moving an array in this study is to form 3D array aperture surrounding multiple sound sources so that the improved spatial resolution in a virtual space can be expected. Numerical simulation was made to examine source localization capabilities of various shapes of array. The 3D beam power maps of hemispherical and spherical distribution are found to have very sharp resolution. For experiments, two sound sources were placed in the middle of defined virtual space and arc-shaped line array was rotated around the sources. It is observed that spherical array show the most accurate determination of multiple sources' positions.

• The Journal of the Acoustical Society of Korea
• /
• 제23권2E호
• /
• pp.56-67
• /
• 2004

Despite abundant research outcomes of blind source separation (BSS) in many types of simulated environments, their performances are still not satisfactory to be applied to the real environments. The major obstacle may seem the finite filter length of the assumed mixing model and the nonlinear sensor noises. This paper presents a two-step speech enhancement method with multiple microphone inputs. The first step performs a frequency-domain BSS algorithm to produce multiple outputs without any prior knowledge of the mixed source signals. The second step further removes the remaining cross-channel interference by a spectral cancellation approach using a probabilistic source absence/presence detection technique. The desired primary source is detected every frame of the signal, and the secondary source is estimated in the power spectral domain using the other BSS output as a reference interfering source. Then the estimated secondary source is subtracted to reduce the cross-channel interference. Our experimental results show good separation enhancement performances on the real recordings of speech and music signals compared to the conventional BSS methods.

• Journal of the Korean Wood Science and Technology
• /
• 제33권4호통권132호
• /
• pp.9-14
• /
• 2005

목재에 대한 탈리그닌처리에 의한 목재의 흡음성능과 구조적 특징의 변화를 관찰하고자 목재횡단방향으로 채취한 원반형시험편을 탈리그닌처리하여 무 처리 목재와 구조적 특징의 변화를 관찰하고, 흡음율을 측정 비교하였다. 측정주파수범위에서 탈리그닌처리한 시험편이 보통의 시험편보다 높은 흡음율을 나타내었으며, 특히 2~4 KHz의 주파수영역에서는 50% 정도 높은 흡음율을 나타내었다. 또한 탈리그닌처리한 목재는 무 처리 목재보다 중량이 8% 정도 감소하였으며, 탈리그닌처리한 목재의 횡단면에서는 다수의 미세할렬이 발생하여 이들이 흡음공극으로 작용하여 특정주파수영역에서 흡음율이 증가된 것으로 사료된다.

• 융합신호처리학회논문지
• /
• 제14권1호
• /
• pp.1-5
• /
• 2013

The two major problems related to the blood flow in replaced prosthetic heart valve are thrombus formation and hemolysis. Reliability of prosthetic valve is very important because its failure means the death of patient. There are many factors affecting the valvular failures and their representatives are mechanical failure and thrombosis, so early noninvasive detection is essentially required. The purpose of this study is to detect the various thromboses formation by using acoustic signal acquisition and its spectral analysis on the frequency domain. We made the thrombosis models using Polydimethylsiloxane (PDMS) and they are thrombosis model on the disc, around the sewing ring and fibrous tissue growth across the orifice of valve. Using microphone and amplifier, we measured the acoustic signal from the prosthetic valve, which is attached to the pulsatile mock circulation system. A/D converter sampled the acoustic signal and the spectral analysis is the main algorithm for obtaining spectrum. Then the spectrum of normal and 5 different kinds of abnormal valve were obtained. Each spectrum waveform shows a primary and secondary peak. The secondary peak changes according to the thrombus model. To quantitatively distinguish the frequency peak of the normal valve from that of the thrombosed valves, analysis using a neural network was employed. Acoustic measurement has been used as a noninvasive diagnostic tool and is thought to be a good method for detecting possible mechanical failure or thrombus.

• 한국소음진동공학회:학술대회논문집
• /
• 한국소음진동공학회 2002년도 춘계학술대회논문집
• /
• pp.346-352
• /
• 2002

This paper is the experimental study to estimate the influence of various design parameters on the performance of mufflers with perforated tubes and through-flow partitions. Muffler types considered in the present work include through-flow chamber, through-flow chamber with partition, and cross-flow chamber. The influences of the design parameters on the performance of the mufflers can be outlined as follows. In the case of the through-f]ow type mufflers, increasing the tube thickness and the hole diameter of the perforated tubes does not change the maximum value of the transmission loss but decrease the cutoff frequency. In the case of the through-flow with partitions type mufflers, it is shown that combining a fe w short chambers and long chambers can modify the frequency locations of the resonance frequencies to optimize the performance of the mufflers. For the case of the cross-flow type mufflers, it is shown that the transmission loss of the mufflers is mainly affected by the lower porosity when the porosities are different in both sides of the plug. Overall, it is shown that performance of the through-flow type with partition type mufflers is excellent in the lower frequency region, where the cross-flow type mufflers have better performance in the higher frequency region.

• Smart Structures and Systems
• /
• 제29권6호
• /
• pp.777-783
• /
• 2022

Even though several deep learning-based methods have been applied in the field of acoustic source localization, the previous works have only been conducted using the two-dimensional representation of the beamforming maps, particularly with the planar array system. While the acoustic sources are more required to be localized in a spherical microphone array system considering that we live and hear in the 3D world, the conventional 2D equirectangular map of the spherical beamforming map is highly vulnerable to the distortion that occurs when the 3D map is projected to the 2D space. In this study, a 3D deep learning approach is proposed to fulfill accurate source localization via distortion-free 3D representation. A target function is first proposed to obtain 3D source distribution maps that can represent multiple sources' positional and strength information. While the proposed target map expands the source localization task into a point-wise prediction task, a PointNet-based deep neural network is developed to precisely estimate the multiple sources' positions and strength information. While the proposed model's localization performance is evaluated, it is shown that the proposed method can achieve improved localization results from both quantitative and qualitative perspectives.