We consider a sensor array located in an enclosure, where arbitrary transfer functions (TFs) relate the source signal and the sensors. The array is used for enhancing a signal contaminated by interference. Constrained minimum power adaptive beamforming, which has been suggested by Frost (1972) and, in particular, the generalized sidelobe canceler (GSC) version, which has been developed by Griffiths and Jim (1982), are the most widely used beamforming techniques. These methods rely on the assumption that the received signals are simple delayed versions of the source signal. The good interference suppression attained under this assumption is severely impaired in complicated acoustic environments, where arbitrary TFs may be encountered. In this paper, we consider the arbitrary TF case. We propose a GSC solution, which is adapted to the general TF case. We derive a suboptimal algorithm that can be implemented by estimating the TFs ratios, instead of estimating the TFs. The TF ratios are estimated by exploiting the nonstationarity characteristics of the desired signal. The algorithm is applied to the problem of speech enhancement in a reverberating room. The discussion is supported by an experimental study using speech and noise signals recorded in an actual room acoustics environment.

Signal enhancement using beamforming and nonstationarity with applications to speech

Speech enhancement and separation are core problems in audio signal processing, with commercial applications in devices as diverse as mobile phones, conference call systems, hands-free systems, or hearing aids. In addition, they are crucial preprocessing steps for noise-robust automatic speech and speaker recognition. Many devices now have two to eight microphones. The enhancement and separation capabilities offered by these multichannel interfaces are usually greater than those of single-channel interfaces. Research in speech enhancement and separation has followed two convergent paths, starting with microphone array processing and blind source separation, respectively. These communities are now strongly interrelated and routinely borrow ideas from each other. Yet, a comprehensive overview of the common foundations and the differences between these approaches is lacking at present. In this paper, we propose to fill this gap by analyzing a large number of established and recent techniques according to four transverse axes: 1 the acoustic impulse response model, 2 the spatial filter design criterion, 3 the parameter estimation algorithm, and 4 optional postfiltering. We conclude this overview paper by providing a list of software and data resources and by discussing perspectives and future trends in the field.

/pdf/a-consolidated-perspective-on-multimicrophone-speech-qnbaqxmtvq.pdf

A Consolidated Perspective on Multimicrophone Speech Enhancement and Source Separation

This paper proposes a new robust adaptive beamformer applicable to microphone arrays. The proposed beamformer is a generalized sidelobe canceller (GSC) with a new adaptive blocking matrix using coefficient-constrained adaptive filters (CCAFs) and a multiple-input canceller with norm-constrained adaptive filters (NCAFs). The CCAFs minimize leakage of the target-signal into the interference path of the GSC. Each coefficient of the CCAFs is constrained to avoid mistracking. The input signal to all the CCAFs is the output of a fixed beamformer. In the multiple-input canceller, the NCAFs prevent undesirable target-signal cancellation when the target-signal minimization at the blocking matrix is incomplete. The proposed beamformer is shown to be robust to target-direction errors as large as 200 with almost no degradation in interference-reduction performance, and it can be implemented with several microphones. The maximum allowable target-direction error can be specified by the user. Simulated anechoic experiments demonstrate that the proposed beamformer cancels interference by over 30 dB. Simulation with real acoustic data captured in a room with 0.3-s reverberation time shows that the noise is suppressed by 19 dB. In subjective evaluation, the proposed beamformer obtains 3.8 on a five-point mean opinion score scale, which is 1.0 point higher than the conventional robust beamformer.

A robust adaptive beamformer for microphone arrays with a blocking matrix using constrained adaptive filters

This book is a companion text to Active Control of Sound by P.A. Nelson and S.J. Elliott, also published by Academic Press. It summarizes the principles underlying active vibration control and its practical applications by combining material from vibrations, mechanics, signal processing, acoustics, and control theory. The emphasis of the book is on the active control of waves in structures, the active isolation of vibrations, the use of distributed strain actuators and sensors, and the active control of structurally radiated sound. The feedforward control of deterministic disturbances, the active control of structural waves and the active isolation of vibrations are covered in detail, as well as the more conventional work on modal feedback. The principles of the transducers used as actuateors and sensors for such control strategies are also given an in-depth description. The reader will find particularly interesting the two chapters on the active control of sound radiation from structures: active structural acoustic control. The reason for controlling high frequency vibration is often to prevent sound radiation, and the principles and practical application of such techniques are presented here for both plates and cylinders. The volume is written in textbook style and is aimed at students, practicing engineers, and researchers.

Active control of vibration, 1st edition

Maximizing the output signal-to-noise ratio (SNR) of a sensor array in the presence of spatially colored noise leads to a generalized eigenvalue problem. While this approach has extensively been employed in narrowband (antenna) array beamforming, it is typically not used for broadband (microphone) array beamforming due to the uncontrolled amount of speech distortion introduced by a narrowband SNR criterion. In this paper, we show how the distortion of the desired signal can be controlled by a single-channel post-filter, resulting in a performance comparable to the generalized minimum variance distortionless response beamformer, where arbitrary transfer functions relate the source and the microphones. Results are given both for directional and diffuse noise. A novel gradient ascent adaptation algorithm is presented, and its good convergence properties are experimentally revealed by comparison with alternatives from the literature. A key feature of the proposed beamformer is that it operates blindly, i.e., it neither requires knowledge about the array geometry nor an explicit estimation of the transfer functions from source to sensors or the direction-of-arrival.

Blind Acoustic Beamforming Based on Generalized Eigenvalue Decomposition

Handsfree speaker input of mobile telephones is most desirable today in order to enable safe operation in cars. This is also a prerequisite for voice control of car devices. This paper presents an "on the road" evaluation of the performance of an adaptive microphone array for speech enhancement in cars. The results show a signal-to-noise ratio (SNR) improvement in the range of 10-15 dB over the telephone frequency range with a slight favour for higher frequencies when driving at normal speeds. >

Adaptive array noise suppression of handsfree speaker input in cars

The problem of controlling the superresolution in adaptive beamformers is treated. A straightforward method is presented that works for both narrowband and broadband arrays. The method is based on forming the blocking matrix in a general sidelobe canceller structure using a spatial FIR filter. The suppression of this spatial filter and the implicit noise of the leaky least-mean-square algorithm together determine the beamformer. >

A spatial filtering approach to robust adaptive beaming

Controlling the resolution in adaptive beamformers is often crucial. A simple method that works for both narrow-band and broad-band arrays is presented. This method is based on the normalized leaky LMS algorithm in conjunction with a generalized sidelobe canceller (GSC) structure, where the GSC is designed using a spatial filtering approach. In essence, the suppression of the spatial filters and the implicit noise of the leaky LMS algorithm together determine the adaptive beamformer. Analytical expressions are given for the Wiener filters and the output spectrum versus frequency and point source location. These expressions are employed in the design specification of the spatial filters and to obtain conditions for a controlled quiescent beamformer response. Simulation results are presented to illustrate the behavior of the array. >

/pdf/adaptive-beamforming-spatial-filter-designed-blocking-matrix-fd7e6prpt8.pdf

Adaptive beamforming: Spatial filter designed blocking matrix

An implementation of a broadband adaptive array on a multiprocessor digital signal processing system for use in a hands free mobile radio telephone is described. This implementation of a five-microphone adaptive Griffiths-Jim array can handle FIR filters with up to 128 taps behind each microphone at a sampling rate of 8 kHz. The filter structure makes it possible to combine an adaptive array with a noise canceler. The near-field problem has been solved by using focusing, a speech-controlled adaptive algorithm, and a short hourglass. Preliminary measurements indicate a considerable potential for this technique in hands-free mobile telephony. The array gives a 20-30 dB suppression of a broadband jammer covering 300-1100 Hz, even with three reflecting walls surrounding the microphone. >

A multi-DSP implementation of a broad-band adaptive beamformer for use in a hands-free mobile radio telephone

The authors treat the problem of finding the Wiener filters for a wideband adaptive beamformer. Explicit expressions are given for the filters and error power spectrum in the frequency domain. The expressions are simple to program and give the possibility of investigating superresolution for wideband signals and of determining sufficient lengths of the finite-impulse-response filters in adaptive arrays. The major results are an analytical expression for the broadband array Wiener solution given for an arbitrary spectrum for target, interference, and noise and an approximate analytic expression for the angular 3-dB beamwidth for broadband signals. >

Ingvar Claesson

Papers

Adaptive array noise suppression of handsfree speaker input in cars

A spatial filtering approach to robust adaptive beaming

Adaptive beamforming: Spatial filter designed blocking matrix

A multi-DSP implementation of a broad-band adaptive beamformer for use in a hands-free mobile radio telephone

The broad-band Wiener solution for Griffiths-Jim beamformers