• Title/Summary/Keyword: Delay and sum beamforming

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Mixed Noise Cancellation by Independent Vector Analysis and Frequency Band Beamforming Algorithm in 4-channel Environments (4채널 환경에서 독립벡터분석 및 주파수대역 빔형성 알고리즘에 의한 혼합잡음제거)

  • Choi, Jae-Seung
    • The Journal of the Korea institute of electronic communication sciences
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    • v.14 no.5
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    • pp.811-816
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    • 2019
  • This paper first proposes a technique to separate clean speech signals and mixed noise signals by using an independent vector analysis algorithm of frequency band for 4 channel speech source signals with a noise. An improved output speech signal from the proposed independent vector analysis algorithm is obtained by using the cross-correlation between the signal outputs from the frequency domain delay-sum beamforming and the output signals separated from the proposed independent vector analysis algorithm. In the experiments, the proposed algorithm improves the maximum SNRs of 10.90dB and the segmental SNRs of 10.02dB compared with the frequency domain delay-sum beamforming algorithm for the input mixed noise speeches with 0dB and -5dB SNRs including white noise, respectively. Therefore, it can be seen from this experiment and consideration that the speech quality of this proposed algorithm is improved compared to the frequency domain delay-sum beamforming algorithm.

Directional headphone design based on delay-weight-sum beamforming technique (지연-가중-합 빔형성 기반의 지향성 헤드폰 설계)

  • Jeong, Jihyeon;Noh, Jeein;Park, Youngjin
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2014.10a
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    • pp.712-712
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    • 2014
  • 본 논문에서는 지연-가중-합 빔형성 방법이 적용된 지향성 헤드폰의 마이크로폰 배치 설계를 설명한다. 마이크로폰의 갯수, 마이크로폰 간의 간격 등이 헤드폰 지향성에 영향을 미치는 설계 변수가 된다. 본 논문에서는 현실성을 고려하여 4개 이하의 마이크로폰을 포함한 10cm 길이의 배열을 타겟으로 한다. 전방으로부터의 소리를 증폭하고 후방으로부터의 소리를 감쇠하여, 전-후방 음압차를 최대화하는 것을 목표로 하였다. 구형 머리전달 함수를 이용한 시뮬레이션을 통해 최적의 마이크로폰 배치를 결정하였다. 설계된 헤드폰은 3개의 마이크로폰을 이용하여 300~3000Hz의 주파수 대역에서 평균 34.6dB의 전-후방 음압차를 보였다. 이 결과는 선행 연구에서 수행된 지연-합 빔형성 방법을 이용한 결과에 비해 8.8dB 뛰어난 성능이다.

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Improvement of Microphone Away Performance in the Low Frequencies Using Modulation Technique (변조 기법을 이용한 마이크로폰 어레이의 저주파 대역 특성 개선)

  • Kim, Gi-Bak;Cho, Nam-Ik
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.42 no.4 s.304
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    • pp.111-118
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    • 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.

Image enhancement in ultrasound passive cavitation imaging using centroid and flatness of received channel data (수신 채널 신호의 무게중심과 평탄도를 이용한 초음파 수동 공동 영상의 화질 개선)

  • Jeong, Mok Kun;Kwon, Sung Jae;Choi, Min Joo
    • The Journal of the Acoustical Society of Korea
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    • v.38 no.4
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    • pp.450-458
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    • 2019
  • Passive cavitation imaging method is used to observe the ultrasonic waves generated when a group of bubbles collapses. A problem with passive cavitation imaging is a low resolution and large side lobe levels. Since ultrasound signals generated by passive cavitation take the form of a pulse, the amplitude distribution of signals received across the receive channels varies depending on the direction of incidence. Both the centroid and flatness were calculated to determine weights at imaging points in order to discriminate between the main and side lobe signals from the signal amplitude distribution of the received channel data and to reduce the side lobe levels. The centroid quantifies how the channel data are distributed across the receive channel, and the flatness measures the variance of the channel data. We applied the centroid weight and the flatness to the passive cavitation image constructed using the delay-and-sum focusing and minimum variance beamforming methods to improve the image quality. Using computer simulation and experiment, we show that the application of weighting in delay-and-sum and minimum variance beamforming reduces side lobe levels.

Microphone Array Based Speech Enhancement Using Independent Vector Analysis (마이크로폰 배열에서 독립벡터분석 기법을 이용한 잡음음성의 음질 개선)

  • Wang, Xingyang;Quan, Xingri;Bae, Keunsung
    • Phonetics and Speech Sciences
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    • v.4 no.4
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    • pp.87-92
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    • 2012
  • Speech enhancement aims to improve speech quality by removing background noise from noisy speech. Independent vector analysis is a type of frequency-domain independent component analysis method that is known to be free from the frequency bin permutation problem in the process of blind source separation from multi-channel inputs. This paper proposed a new method of microphone array based speech enhancement that combines independent vector analysis and beamforming techniques. Independent vector analysis is used to separate speech and noise components from multi-channel noisy speech, and delay-sum beamforming is used to determine the enhanced speech among the separated signals. To verify the effectiveness of the proposed method, experiments for computer simulated multi-channel noisy speech with various signal-to-noise ratios were carried out, and both PESQ and output signal-to-noise ratio were obtained as objective speech quality measures. Experimental results have shown that the proposed method is superior to the conventional microphone array based noise removal approach like GSC beamforming in the speech enhancement.

Development of Photoacoustic System for Breast Cancer Detection (유방암 진단용 광음향 영상 시스템 개발)

  • Lee, Soonhyouk;Ji, Yun-Seo;Lee, Rena
    • Progress in Medical Physics
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    • v.24 no.3
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    • pp.183-190
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    • 2013
  • Recently, the photoacoustic imaging system has been widely and intensively developed, and has been shown the possibility of diagnosis for early stage cancer. In this study, we developed a photoacoustic tomography imaging system with a commercial ultra sound device and a linear array probe. A tube phantom and a chicken breast phantom was made for the possibility of a system as a breast cancer detection. A moving average filter and a band pass filter with 3~6 MHz bandwidth were developed for background noise elimination before delay-and-sum beamforming algorithm was used for image reconstruction. As a result, we showed that some signal processing procedure before beamforming was effective for the photoacoustic image reconstruction.

Designing a Microphone Array System for Noise Measurements on High-Speed Trains (고속철도 차량의 소음 측정을 위한 마이크로폰 어레이 설계에 관한 연구)

  • No, Hui-Min;Choe, Seong-Hun;Hong, Seok-Yun
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2011.10a
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    • pp.717-722
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    • 2011
  • In this paper, noise source localization of the Korean high speed train was conducted by using delay and sum beam-forming method of a microphone array. At first, the microphone array having irregular configuration was designed and the resolution of which was analyzed from parameters such as 3-dB bandwidth and maximum side-lobe level. After the demonstration, the microphone array was applied on the high speed train and noise localization of the high speed train driving at 300 km/h was performed successfully.

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Ultrasonic Source Localization and Visualization Technique for Fault Detection of a Power Distribution Equipment (배전설비 결함 검출을 위한 초음파 음원 위치추정 및 시각화 기법)

  • Park, Jin Ha;Jung, Ha Hyoung;Lyou, Joon
    • Journal of Institute of Control, Robotics and Systems
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    • v.21 no.4
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    • pp.315-320
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    • 2015
  • This paper describes the implemenation of localization and visualization scheme to find out an ultrasonic source caused by defects of a power distribution line equipment. To increase the fault detection performance, $2{\times}4$ sensor array is configured with MEMS ultrasonic sensors, and from the sensor signals aquired, the azimuth and elevation angles of the ultrasonic source is estimated based on the delay-sum beam forming method. Also, to visualize the estimated location, it is marked on the background image. Experimental results show applicability of the present technique.

Impulsive sound localization using crest factor of the time-domain beamformer output (빔형성기 출력의 파고율을 이용한 충격음의 방향 추정)

  • Seo, Dae-Hoon;Choi, Jung-Woo;Kim, Yang-Hann
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2014.10a
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    • pp.713-717
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    • 2014
  • This paper presents a beamforming technique for locating impulsive sound source. The conventional frequency-domain beamformer is advantageous for localizing noise sources for a certain frequency band of concern, but the existence of many frequency components in the wide-band spectrum of impulsive noise makes the beamforming image less clear. In contrast to a frequency-domain beamformer, it has been reported that a time-domain beamformer can be better suited for transient signals. Although both frequency- and time-domain beamformers produce the same result for the beamforming power, which is defined as the RMS value of its output, we can use alternative directional estimators such as the peak value and crest factor to enhance the performance of a time-domain beamformer. In this study, the performance of three different directional estimators, the peak, crest factor and RMS output values, are investigated and compared with the incoherent interfering noise embedded in multiple microphone signals. The proposed formula is verified via experiments in an anechoic chamber using a uniformly spaced linear array. The results show that the peak estimation of beamformer output determines the location with better spatial resolution and a lower side lobe level than crest factor and RMS estimation in noise free condition, but it is possible to accurately estimate the direction of the impulsive sound source using crest factor estimation in noisy environment with stationary interfering noise.

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