We study a time domain decorrelation method of source signal separation from convolutive sound mixtures based on an infinite impulse response （IIR） model. The IIR model uses fewer parameters to capture the physical ...We study a time domain decorrelation method of source signal separation from convolutive sound mixtures based on an infinite impulse response （IIR） model. The IIR model uses fewer parameters to capture the physical mixing process and is useful for finding low dimensional separating solutions. We present inversion formulas to decorrelate the mixture signals and derive filter equations involving second order time lagged statistics of mixtures. We then formulate an 11 constrained minimization problem and solve it by an iterative method. Numerical experiments on recorded sound mixtures show that our method is capable of sound separation in low dimensional parameter spaces with good perceptual quality and low correlation coefficient comparable to the known infomax method.展开更多
It is well known that the performance of conventional adaptive beamformers degrades severely due to the presence of coherent or correlated interferences(multipath propagation) and various techniques have been develope...It is well known that the performance of conventional adaptive beamformers degrades severely due to the presence of coherent or correlated interferences(multipath propagation) and various techniques have been developed to improve the performance of the beamformer.However,most of the work in the past has been focused on the narrowband case.In this paper,the wideband beamforming problem in the presence of multipath signals is addressed,with a novel approach proposed by employing a pre-processing stage based on the frequency invariant beamforming(FIB) technique.In this approach,the received wideband array signals are first processed by an FIB network,and then a traditional narrowband adaptive beamformer or an appropriate instantaneous blind source separation(BSS) algorithm can be applied to the network outputs.It is shown that with the proposed structure,cancellation of the desired signal is reduced,leading to a significantly improved output signal to interference plus noise ratio(SINR).展开更多
基金partially supported by NSF grants DMS-0712881, NIH grant 2R44DC006734the CORCLR (Academic Senate Council on Research, Computing and Library Resources) faculty research grant MI-2006-07-6, and a Pilot award of the Center for Hearing Research at UC Irvine
文摘We study a time domain decorrelation method of source signal separation from convolutive sound mixtures based on an infinite impulse response （IIR） model. The IIR model uses fewer parameters to capture the physical mixing process and is useful for finding low dimensional separating solutions. We present inversion formulas to decorrelate the mixture signals and derive filter equations involving second order time lagged statistics of mixtures. We then formulate an 11 constrained minimization problem and solve it by an iterative method. Numerical experiments on recorded sound mixtures show that our method is capable of sound separation in low dimensional parameter spaces with good perceptual quality and low correlation coefficient comparable to the known infomax method.
文摘It is well known that the performance of conventional adaptive beamformers degrades severely due to the presence of coherent or correlated interferences(multipath propagation) and various techniques have been developed to improve the performance of the beamformer.However,most of the work in the past has been focused on the narrowband case.In this paper,the wideband beamforming problem in the presence of multipath signals is addressed,with a novel approach proposed by employing a pre-processing stage based on the frequency invariant beamforming(FIB) technique.In this approach,the received wideband array signals are first processed by an FIB network,and then a traditional narrowband adaptive beamformer or an appropriate instantaneous blind source separation(BSS) algorithm can be applied to the network outputs.It is shown that with the proposed structure,cancellation of the desired signal is reduced,leading to a significantly improved output signal to interference plus noise ratio(SINR).
