• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 2017.11a
• /
• pp.44-45
• /
• 2017

본 논문에서는 비균등 선형 마이크로폰 어레이를 활용한 convolutional neural network (CNN) 기반의 음원분리 방법을 제안한다. 우선, 주어진 어레이 배치에 따라 채널간의 시간차를 분석하고, 분석된 시간차에 따라 주파수별로 방사각과 넓이에 따라 입력 오디오 신호의 spectral magnitude를 예측한다. 그러고 나서, CNN 분류기로부터 최적의 방사각과 넓이를 선별하고 이를 통해 음원을 분리한다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 2015.11a
• /
• pp.21-22
• /
• 2015

현재 UHDTV (Ultra-High-Definition TV) 시대에 사용될 멀티미디어 부호화로 MPEG-H를 표준화로 진행하고 있다. 향후 방송용 오디오 콘텐츠는 채널 오디오 콘텐츠에서 진화하여 객체 오디오 콘텐츠까지도 필요하게 된다. 이에 따라, 본 논문에서는 고품질의 방송용 오디오 콘텐츠를 제작하기 위한 비균등 선형 마이크로폰 어레이 기반의 음원분리 방법을 제안한다. 제안된 방법은 주어진 어레이 배치에 따라 채널간의 시간차를 분석하고, 이에 따른 객체 오디오 생성을 위한 음원분리 기술을 적용한다. 제안된 기법의 성능을 검증하기 위하여 음원분리도를 측정하였고, MVDR (Minimum Variance Distortionless Response) 빔형성기와 성능을 비교하였다. 비교 결과, 제안된 기법이 MVDR 빔형성기에 비하여 12.8% 높은 음원분리도 수치를 나타낸 것을 확인하였다.

• The Journal of Engineering Research
• /
• v.8 no.1
• /
• pp.59-65
• /
• 2006

본 논문에서는 마이크로폰 어레이 시스템을 이용하여 여러 화자의 음성 정보로부터 각 화자가 위치한 방향을 추정하는 기술 개발 내용을 다룬다. 성능 향상을 위한 전처리 과정으로 비선형 증폭기를 사용하여 거리에 따른 영향을 최소화하는 과정과 잡음에 대한 강인성을 얻기 위해 음성활성 영역을 검출하는 과정을 포함한다. 등간격으로 배치된 마이크로폰 어레이 시스템의 기하학적 특성에 따른 음원의 위치와 신호의 지연시간차이와의 상관관계로부터 화자의 위치를 역으로 추정하는 알고리즘을 기본으로 하여 가능성 척도를 계산하고 이를 활용하여 가능성이 높은 것들을 클러스터링하여 가능성이 있는 후보를 선정하여 화자의 방향을 검지한다. 이 과정에서 오인식을 최소화하기 위하여 가능성이 희박한 영역에 대한 추정 억제 방법으로 부정식 추론법을 적용하였다. 2 화자의 음성 신호를 입력으로 한 실험을 통하여 제안한 방법에 의한 다화자 방향검지의 가능성을 알아보았다.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.42 no.4 s.304
• /
• pp.111-118
• /
• 2005

In this paper, we employ the modulation technique for improving the characteristics of beamformer in the low frequencies and thus improving the overall noise reduction performance. In the 1-dimensional uniform linear microphone arrays, we can suppress the narrowband noise component using the delay-and-sum beamforming. But, for the wideband noise signal, the delay-and-sum beamformer does not work well for the reduction of low frequency component because the inter-element spacing is usually set to avoid spatial aliasing at high frequencies. Hence, the beamwidth is not uniform with respect to each frequency and it is usually wider at the low frequencies. In order to obtain the beamwidth independent of frequencies, subarray systems[1][2][3][4] and multi-beamforming[5] have been proposed. However these algorithms need large space and more microphones since they are based on the theory that the size of the array is proportional to the wavelength of the input signal. In the proposed beamformer, we reduce the low frequency noise by using modulation technique that does not need additional sensors or non-uniform spacing. More Precisely, the array signals are split into subbands, and the low frequency components are shifted to high frequencies by modulation and reduced by the delay-and-sum beamforming techniques with small size microphone array. Experimental results show that the proposed technique Provides better performance than the conventional ones, especially in the low frequency band.

• Journal of Broadcast Engineering
• /
• v.23 no.4
• /
• pp.519-525
• /
• 2018

This paper deals with noise suppression of multi-channel data captured by microphone array using multi-channel Wiener filter. Multi-channel Wiener filter does not rely on information about the direction of the target speech and can be partitioned into an MVDR (Minimum Variance Distortionless Response) spatial filter and a single channel spectral filter. The acoustic transfer function between the single speech source and microphones can be estimated by subspace decomposition of multi-channel Wiener filter. The errors are incurred in the estimation of the acoustic transfer function due to the errors in the estimation of correlation matrices, which in turn results in speech distortion in the MVDR filter. To alleviate the speech distortion in the MVDR filter, diagonal loading is applied. In the experiments, database with seven microphones was used and MFCC distance was measured to demonstrate the effectiveness of the diagonal loading.

• The Journal of the Acoustical Society of Korea
• /
• v.34 no.1
• /
• pp.60-65
• /
• 2015

In this paper, we used the sound of gunshots recorded by multiple microphones to increase the accuracy of the calculation of the distance between sniper and the microphone array. This method is crucial for achieving military objectives. Gunshots are comprised of the explosion of driving gas from the muzzle and the supersonic shock wave from the flying bullet. The original distance calculation method compares the time difference of arrival and angle of incidence to estimate the sniper's location. The disadvantage of this method is that when the angles of incidence coincide the margin of error increases, to solve this problem we suggest a new method using the characteristic changes of the shock wave with the increase of perpendicular distance between the microphone and the trajectory of the bullet. This theory is verified by experiments.

• Journal of Broadcast Engineering
• /
• v.21 no.2
• /
• pp.169-179
• /
• 2016

Recently, MPEG-H has been standardizing for a multimedia coder in UHDTV (Ultra-High-Definition TV). Thus, the demand for not only channel-based audio contents but also object-based audio contents is more increasing, which results in developing a new technique of converting channel-based audio contents to object-based ones. In this paper, a non-uniform linear microphone array based source separation method is proposed for realizing such conversion. The proposed method first analyzes the arrival time differences of input audio sources to each of the microphones, and the spectral magnitudes of each sound source are estimated at the horizontal directions based on the analyzed time differences. In order to demonstrate the effectiveness of the proposed method, objective performance measures of the proposed method are compared with those of conventional methods such as an MVDR (Minimum Variance Distortionless Response) beamformer and an ICA (Independent Component Analysis) method. As a result, it is shown that the proposed separation method has better separation performance than the conventional separation methods.

• Proceedings of the Acoustical Society of Korea Conference
• /
• spring
• /
• pp.403-406
• /
• 2000

파라메트릭 송파 방식을 공기중에서의 음향 변환기에 적용시키기 위한 검토로써, 비교적 공진 주파수가 낮은 기중 초음파 발생 소자를 이용하여 파라메트릭 송파기를 제작하였다. 실험에서는 송파기로서 공진주파수 38.6kHz를 갖는 음원을 제작하여 이용하였다. 비선형 왜곡으로부터 파라메트릭 어레이에 의한 차주 파수가 발생함을 알아보기 위하여 1kHz의 차주파수를 갖도록 38.1kHz와 39.1kHz의 1차파를 신호발생기로부터 인가하고 마이크로폰을 사용하여 수신하였다. 공기중에서 파라메트릭 효과의 확인을 통하여 파라메트릭 송파기에서 방사되는 1차파와 전파경로중에서 생성하는 2차파의 거동을 이론면에서 고찰하고, 실험 결과와의 비교 검토를 행하여 기중 파라메트릭 송파기 개발의 가능성을 검토하여 보았다.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
• /
• 1996.04a
• /
• pp.291-297
• /
• 1996

To visualize sound field or to identify noise sources, we can use many methods such as intensity method, acoustic holographic method, source identification method using line array, etc. Conventionally all these methods are performed with the assumption of stationary condition in space and time. But for moving source, spatial characteristics and frequency components are changing, so we need another processing algorithm. This paper shows some experimental results - sound field by moving noise sources. In the experiment cross type microphone line array is used for sensing pressure and cars and a motorcycle are used as moving sources that are assumed to have constant speed. The processing methods are acoustic holographic method, spherical beamforming and spectrogram.

• Journal of KSNVE
• /
• v.6 no.6
• /
• pp.835-842
• /
• 1996

To identify the locations and strengths of acoustic sources, one may use a microphone line array. Apparent advantage of the source identification method utilizing a line array is that it requires less measurement points than intensity method and holography. This method is based on the information of magnitude and phase difference between pressure signals at each microphone. Since those differences are dependent on the source model, we have to assume them such as plane, monopole, etc. In this paper the conventional source identification methods such as beamforming method and MUSIC method are briefly reviewed by modeling a source as plane and spherical wave, then a modified method is introduced. This can be applied to sound field which may by either coherent or incoherent. Typical simulations and experiment are performed to confirm this identification method.