• Title/Summary/Keyword: MVDR

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A Microphone Array Beamformer for the Performance Enhancement of Speech Recognizer in Car (차량환경에서 음성인식 성능 향상을 위한 마이크로폰 어레이 빔형성 기법)

  • Han Chul-Hee;Kang Hong-Goo;Hwang Youngsoo;Youn Dae-Hee
    • The Journal of the Acoustical Society of Korea
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    • v.24 no.7
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    • pp.423-430
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    • 2005
  • In this paper. a microphone array beamforming algorithm that reduces the signal distortion caused by reverberation and near-field effect in car environment is proposed. When reverberation or near-field effect is present, an optimum beamformer should be constructed with a steering vector consisting of transfer functions between source and microphones, but it is generally difficult to estimate transfer functions on-line without knowledge of the source signal. Instead, a sub-optimal beamforming algorithm that reduces signal distortion is proposed. It is constructed with steering vectors consisting of relative transfer functions between reference sensor and other sensors. In order to evaluate the performance of the proposed algorithm. we had recorded noisy speech database in a car, and performed speech recognition experiments with HMM Toolkit (HTK) released by Cambridge University. The recognition rate of the proposed algorithm was 15 percents higher than that of the conventional far-field beamformers in best case.

A Calibration Technique for Array antenna based GPS Receivers (배열 안테나 기반 GPS 수신기에서의 교정 방안)

  • Kil, Haeng-bok;Joo, Hyun;Lee, Chulho;Jeong, Eui-Rim
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.22 no.4
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    • pp.683-690
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    • 2018
  • In this paper, a new signal processing technique is proposed for calibrating gain, phase, delay offsets in array antenna based anti-jamming minimum variance distortionless response (MVDR) global-positioning-system (GPS) receivers. The proposed technique estimates gain, phase and delay offsets across the antennas, and compensates for the offsets based on the estimates. A pilot signal with good correlation characteristics is used for accurate estimation of the gain, phase and delay offsets. Based on the cross-correlation, the delay offset is first estimated and then gain/phase offsets are estimated. For fine delay offset estimation and compensation, an interpolation technique is used, and specifically, the discrete Fourier transform (DFT) is employed for the interpolation technique to reduce the computational complexity. The proposed technique is verified through computer simulation using MATLAB. According to the simulation results, the proposed technique can reduce the gain, phaes and delay offset to 0.01 dB, 0.05 degree, and 0.5 ns, respectively.

Adaptive Beamforming System Architecture Based on AOA Estimator (AOA 추정기 기반의 적응 빔형성 시스템 구조)

  • Mun, Ji-Youn;Bae, Young-Chul;Hwang, Suk-Seung
    • The Journal of the Korea institute of electronic communication sciences
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    • v.12 no.5
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    • pp.777-782
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    • 2017
  • The Signal Intelligence (SIGINT) system based on the adaptive beamformer, comprised of the AOA estimator followed by the interference canceller, is a cutting edge technology for collecting various signal information utilizing all sorts of devices such as the radar and satellite. In this paper, we present the efficient adaptive SIGINT structure consisted of an AOA estimator and an adaptive beamformer. For estimating AOA information of various signals, we employ the Multiple Signal Classification (MUSIC) algorithm and for efficiently suppressing high-power interference signals, we employ the Minimum Variance Distortionless Response (MVDR) algorithm. Also, we provide computer simulation examples to verify the performance of the presented adaptive beamformer structure.

Single-Channel Non-Causal Speech Enhancement to Suppress Reverberation and Background Noise

  • Song, Myung-Suk;Kang, Hong-Goo
    • The Journal of the Acoustical Society of Korea
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    • v.31 no.8
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    • pp.487-506
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    • 2012
  • This paper proposes a speech enhancement algorithm to improve the speech intelligibility by suppressing both reverberation and background noise. The algorithm adopts a non-causal single-channel minimum variance distortionless response (MVDR) filter to exploit an additional information that is included in the noisy-reverberant signals in subsequent frames. The noisy-reverberant signals are decomposed into the parts of the desired signal and the interference that is not correlated to the desired signal. Then, the filter equation is derived based on the MVDR criterion to minimize the residual interference without bringing speech distortion. The estimation of the correlation parameter, which plays an important role to determine the overall performance of the system, is mathematically derived based on the general statistical reverberation model. Furthermore, the practical implementation methods to estimate sub-parameters required to estimate the correlation parameter are developed. The efficiency of the proposed enhancement algorithm is verified by performance evaluation. From the results, the proposed algorithm achieves significant performance improvement in all studied conditions and shows the superiority especially for the severely noisy and strongly reverberant environment.

Steering Angle Error Compensation Algorithm Appropriate for Rapidly Moving Sources (빠른 속도로 기동하는 표적 환경에 적합한 조향각 오차 보정기법)

  • 박규태;박도현;이정훈;이균경
    • The Journal of the Acoustical Society of Korea
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    • v.23 no.3
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    • pp.206-213
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    • 2004
  • This paper presents a steering angle error compensation (SAEC) algorithm that is appropriate for rapidly moving sources. The Proposed algorithm utilizes a modal covariance matrix from multiple frequency components instead of the multiple snapshots in a narrowband SAEC, and estimates the steering error by maximizing the wideband WVDR output power using a first-order Taylor series approximation of the modal steering vector in terms of the steering error. As such, the steering error can be compensated with short observation times. Several simulations using artificial and sea trial data are used to demonstrate the Performance of the proposed algorithm.

Performance Analysis of TPMS Beamformer According to Variance of Antenna Interelement Spacing (안테나 간격 변화에 대한 TPMS 빔형성기 성능분석)

  • Choi, Byung-Sang;Kim, Seong-Min;Hwang, Suk-Seung
    • The Journal of the Korea institute of electronic communication sciences
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    • v.8 no.6
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    • pp.907-915
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    • 2013
  • Tire Pressure Monitoring System (TPMS) is an auxiliary safety system for recognizing the condition of tires based on the pressure and temperature data transmitted from the sensor unit installed on a tire of the vehicle. Using TPMS, a driver can frequently check the state of tires and it aids to maintain the optimum running condition of the vehicle. Since TPMS must utilize the wireless communication technique to transmit data from a sensor unit to a signal processing unit installed in the vehicle, it suffers from interference signals caused by various external electrical or electronic devices. In order to suppress high-power interference signals, we employ beamforming techniques based on the uniform linear antenna array. As the number of the antennas is increased, the performance of the interference suppression is improved. However, there is the limit of the number of antennas, installed in the center of a vehicle, because of its size. In this paper, we compare and analyze the performance of the beamformer, when reducing the interelement spacing of antennas, to increase the number of the receiving antennas. For the performance analysis of the beamformers, we consider the switching beamformer and minimum-variance distortionless-response (MVDR) beamformer for TPMS, recently proposed.

Using The Matrix Pencil Method to Frequency Estimate Algorithm of OFDM System (Matrix Pencil Method를 이용한 OFDM의 주파수 추정)

  • 차정근;강석진;박상백;고진환
    • Proceedings of the Korean Information Science Society Conference
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    • 2003.04d
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    • pp.73-75
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    • 2003
  • OFDM 전송방식에 있어서 중요한 주파수 옵셋 추정을 하는데 있어서 기존의 FFT 방법이 가지고 있는 문제점을 보완하는 알고리즘이 많이 연구되고 있다. FFT의 정수배 옵셋외에 소수배 옵셋이 생겼을때 제대로 추정해 낼 수 없는 점을 보완하는 High resolutional technical 알고리즘을 보면 MVDR, MUSIC, root MUSIC, PISARENCO 등이 있다. 본 논문에서는 이러한 알고리즘 중에 MPM(Matrix Pencil Method)를 이용하여 FFT의 문제점을 보완하고 옵셋 추정을 시뮬레이션 해 보았다.

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Analysis on performance of grid-free compressive beamforming based on experiment (실험 기반 무격자 압축 빔형성 성능 분석)

  • Shin, Myoungin;Cho, Youngbin;Choo, Youngmin;Lee, Keunhwa;Hong, Jungpyo;Kim, Seongil;Hong, Wooyoung
    • The Journal of the Acoustical Society of Korea
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    • v.39 no.3
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    • pp.179-190
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    • 2020
  • In this paper, we estimated the Direction of Arrival (DOA) using Conventional BeamForming (CBF), adaptive beamforming and compressive beamforming. Minimum Variance Distortionless Response (MVDR) and Multiple Signal Classification (MUSIC) are used as the adaptive beamforming, and grid-free compressive sensing is applied for the compressive sensing beamforming. Theoretical background and limitations of each technique are introduced, and the performance of each technique is compared through simulation and real experiments. The real experiments are conducted in the presence of reflected signal, transmitting a sound using two speakers and receiving acoustic data through a linear array consisting of eight microphones. Simulation and experimental results show that the adaptive beamforming and the grid-free compressive beamforming have a higher resolution than conventional beamforming when there are uncorrelated signals. On the other hand, the performance of the adaptive beamforming is degraded by the reflected signals whereas the grid-free compressive beamforming still improves the conventional beamforming resolution regardless of reflected signal presence.

Sound Source Separation Using Interaural Intensity Difference in Closely Spaced Stereo Omnidirectional Microphones (인접 배치된 스테레오 무지향성 마이크로폰 환경에서 양이간 강도차를 활용한 음원 분리 기법)

  • Chun, Chan Jun;Jeong, Seok Hee;Kim, Hong Kook
    • Journal of the Institute of Electronics and Information Engineers
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    • v.50 no.12
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    • pp.191-196
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    • 2013
  • In this paper, the interaural intensity difference (IID)-based sounr source separation method in closely spaced stereo omnidirectional microphones is proposed. First, in order to improve the channel separability, a minimum variance distortionless response (MVDR) beamformer is employed to increase the intensity difference between stereo channels. After that, IID-based sound source separation method is applied. In order to evaluate the performance of the proposed method, source-to-distortion ratio (SDR), source-to-interference ratio (SIR), and sources-to-artifacts ratio (SAR), which are defined as objective evaluation criteria in stereo audio source separation evaluation campaign (SASSEC), are measured. As a result, it was shown from the objective evaluation that the proposed method outperforms a sound source separation method without applying a beamformer.

Performance Evaluation of Satellite System Based on Transmission Beamformer (송신 빔형성기 기반의 위성 시스템 구조 성능평가)

  • Mun, Ji-Youn;Hwang, Myeong-Hwan;Hwang, Suk-Seung
    • The Journal of the Korea institute of electronic communication sciences
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    • v.13 no.4
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    • pp.713-720
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    • 2018
  • The Signal Intelligence (SIGINT) system based on Angle-of-Arrival(AOA) estimation, interference suppression, and transmission beamforming techniques is a cutting edge technology for efficiently collecting various signal information. In this paper, we present the efficient structure of a satellite system consisted of an AOA estimator, an adaptive beamformer, a signal processing and D/B unit, and a transmission beamformer, for collecting signal information. For accurately estimating AOAs of various signals, efficiently suppressing interference or jamming signals, and efficiently transmitting the collected information or data, we employ Multiple Signal Classification (MUSIC), Minimum Variance Distortionless Response (MVDR), and Minimum Mean Square Error (MMSE) algorithms, respectively. Also, we evaluate and analysis the performance of the presented satellite system through the computer simulation.