Controlling Speech Distortion in Adaptive Frequency-domain Principal Eigenvector Beamforming

نویسندگان

  • Ernst Warsitz
  • Reinhold Haeb-Umbach
چکیده

Broadband adaptive beamformers, which use a narrowband SNRmaximization optimization criterion for noise reduction, typically cause distortions of the desired speech signal at the beamformer output. In this paper two methods are investigated to control the speech distortion by comparing the eigenvector beamformer with a maximum likelihood beamformer: One is an analytic solution for the ideal case of absence of reverberation and the other one is a statistically motivated approach. We use the recently introduced gradient-ascent algorithm for adaptive principal eigenvector beamforming and then normalize the filter coefficients by the proposed distortion control methods. Experimental results in terms of the achievable SNR gain and a perceptual speech quality measure are given for the normalized eigenvector beamformer and are compared to standard beamforming methods.

برای دانلود متن کامل این مقاله و بیش از 32 میلیون مقاله دیگر ابتدا ثبت نام کنید

ثبت نام

اگر عضو سایت هستید لطفا وارد حساب کاربری خود شوید

منابع مشابه

Blind adaptive principal eigenvector beamforming for acoustical source separation

For separating multiple speech signals given a convolutive mixture, time-frequency sparseness of the speech sources can be exploited. In this paper we present a multi-channel source separation method based on the concept of approximate disjoint orthogonality of speech signals. Unlike binary masking of singlechannel signals as e.g. applied in the DUET algorithm we use a likelihood mask to contro...

متن کامل

A Novel Frequency Domain Linearly Constrained Minimum Variance Filter for Speech Enhancement

A reliable speech enhancement method is important for speech applications as a pre-processing step to improve their overall performance. In this paper, we propose a novel frequency domain method for single channel speech enhancement. Conventional frequency domain methods usually neglect the correlation between neighboring time-frequency components of the signals. In the proposed method, we take...

متن کامل

Optimal and Adaptive Subband Beamforming Principles and Applications

This part discusses signal processing methods for speech extraction in use with voice communication applications such as personal digital assistants (PDA:s), mobile telephone terminals and personal computers. The speaker will be distant from the device and thus the speech signal entering the device will be subject to reverberation as well as disturbed by background noise. Further more, the comp...

متن کامل

Speech enhancement using improved generalized sidelobe canceller in frequency domain with multi-channel postfiltering

In this paper, we propose a speech enhancement algorithm which has the feature of interaction between adaptive beamforming and multi-channel postfilter. A novel subband feedback controller based on speech presence probability is applied to Generalized Sidelobe Canceller algorithm to obtain a more robust adaptive beamforming in adverse environment and alleviate the problem of signal cancellation...

متن کامل

Adaptive beamforming and soft missing data decoding for robust speech recognition in reverberant environments

This paper presents a novel approach to combine microphone array processing and robust speech recognition for reverberant multi-speaker environments. Spatial cues are extracted from a microphone array and automatically clustered to estimate localization masks in the time-frequency domain. The localization masks are then used to blindly design adaptive filters in order to enhance the source sign...

متن کامل

ذخیره در منابع من


  با ذخیره ی این منبع در منابع من، دسترسی به آن را برای استفاده های بعدی آسان تر کنید

برای دانلود متن کامل این مقاله و بیش از 32 میلیون مقاله دیگر ابتدا ثبت نام کنید

ثبت نام

اگر عضو سایت هستید لطفا وارد حساب کاربری خود شوید

عنوان ژورنال:

دوره   شماره 

صفحات  -

تاریخ انتشار 2006