Ken'ichi Furuya

Robust Speech Dereverberation Using Multichannel Blind Deconvolution With Spectral Subtraction

A robust dereverberation method is presented for speech enhancement in a situation requiring adaptation where a speaker shifts his/her head under reverberant conditions causing the impulse responses to change frequently. We combine correlation-based blind deconvolution with modified spectral subtraction to improve the quality of inverse-filtered speech degraded by the estimation error of inverse filters obtained in practice. Our method computes inverse filters by using the correlation matrix between input signals that can be observed without measuring room impulse responses. Inverse filtering reduces early reflection, which has most of the power of the reverberation, and then, spectral subtraction suppresses the tail of the inverse-filtered reverberation. The performance of our method in adaptation is demonstrated by experiments using measured room impulse responses. The subjective results indicated that this method provides superior speech quality to each of the individual methods: blind deconvolution and spectral subtraction.

Method and apparatus for dereverberation

In a method and an apparatus for dereverberation provided speech from speaker is received by a first and second channel microphones which are disposed at different locations, and is input to a first and a second channel reverberant speech input terminal. Input signal in each channel is processed by an inverse filter processor, and a dereverberation evaluation part evaluates dereverberation performance on the basis of an output signal from the inverse filter processor and the input signals of respective channels. Subsequently, filter coefficients in the inverse filter processor are determined and updated in accordance with the evaluation so that a result of evaluation is brought closer to an optimum. By repeating this process, an optimum dereverberation is always enabled in a manner following a variation in an in-room impulse response.

Estimating Direct-to-Reverberant Energy Ratio Using D/R Spatial Correlation Matrix Model

We present a method for estimating the direct-to-reverberant energy ratio (DRR) that uses a direct and reverberant sound spatial correlation matrix model (Hereafter referred to as the spatial correlation model). This model expresses the spatial correlation matrix of an array input signal as two spatial correlation matrices, one for direct sound and one for reverberation. The direct sound propagates from the direction of the sound source but the reverberation arrives from every direction uniformly. The DRR is calculated from the power spectra of the direct sound and reverberation that are estimated from the spatial correlation matrix of the measured signal using the spatial correlation model. The results of experiment and simulation confirm that the proposed method gives mostly correct DRR estimates unless the sound source is far from the microphone array, in which circumstance the direct sound picked up by the microphone array is very small. The method was also evaluated using various scales in simulated and actual acoustical environments, and its limitations revealed. We estimated the sound source distance using a small microphone array, which is an example of application of the proposed DRR estimation method.

Analytical Approach to Wave Field Reconstruction Filtering in Spatio-Temporal Frequency Domain

For transmission of a physical sound field in a large area, it is necessary to transform received signals of a microphone array into driving signals of a loudspeaker array to reproduce the sound field. We propose a method for transforming these signals by using planar or linear arrays of microphones and loudspeakers. A continuous transform equation is analytically derived based on the physical equation of wave propagation in the spatio-temporal frequency domain. By introducing spatial sampling, the uniquely determined transform filter, called a wave field reconstruction filter (WFR filter), is derived. Numerical simulations show that the WFR filter can achieve the same performance as that obtained using the conventional least squares (LS) method. However, since the proposed WFR filter is represented as a spatial convolution, it has many advantages in filter design, filter size, computational cost, and filter stability over the transform filter designed by the LS method.

Reproducing Virtual Sound Sources in Front of a Loudspeaker Array Using Inverse Wave Propagator

It has been possible to reproduce point sound sources between listeners and a loudspeaker array by using the focused-source method. However, this method requires physical parameters of the sound sources to be reproduced, such as source positions, directions, and original signals. This fact makes it difficult to apply the method to real-time reproduction systems because decomposing received signals into such parameters is not a trivial task. This paper proposes a method for recreating virtual sound sources in front of a planar or linear loudspeaker array. The method is based on wave field synthesis but extended to include the inverse wave propagator often used in acoustical holography. Virtual sound sources can be placed between listeners and a loudspeaker array even when the received signals of a microphone array equally aligned with the loudspeaker array are only known. Numerical simulation results are presented to compare the proposed and focused-source methods. A system was constructed consisting of linear microphone and loudspeaker arrays and measurement experiments were conducted in an anechoic room. When comparing the sound field reproduced using the proposed method with that using the focused-source method, it was found that the proposed method could reproduce the sound field at almost the same accuracy.

Speech Dereverberation by Combining Mint-Based Blind Deconvolution and Modified Spectral Subtraction

A dereverberation technique is developed to provide an alternative means of reducing reverberation in speech signals. The conventional MINT (the multiple-input/output inverse-filtering theorem) method uses the room impulse responses to calculate the inverse filters, so it cannot recover speech signals in practice, where the room impulse responses are unknown in advance. Our method blindly estimates the inverse filters by computing the correlation matrix between input signals that can be observed, instead of room impulse responses. We also combine the inverse filtering with modified spectral subtraction against the estimation error of inverse filters used in the field. The performance of the proposed method is demonstrated using actual room impulse responses

Diffused Sensing for Sharp Directive Beamforming

We generalized our previously proposed diffused sensing for a microphone array design to achieve sharp directive beamforming to enable various filter design methods to be applied. In the conventional microphone array, various filter design methods have been studied to narrow the directivity beam width. However, it is difficult to minimize the power of interference sources in the beamforming output (output interference power) over a broad frequency range since the cross-correlation between transfer functions from sound sources to microphones increases in some frequencies. With the diffused sensing, the cross-correlation is minimized by physically varying the transfer functions. We investigated how a microphone array should be designed in order to minimize the cross-correlation between transfer functions and found that placing the array in a diffuse acoustic field produces optimum results. Because the transfer functions are known a priori, this finding makes it possible to narrow the directivity beam width over a broad frequency range. This observation can be practically achieved by placing microphones inside a reflective enclosure, part of which is open to let sound waves enter. We conducted experiments using 24 microphones and confirmed that the output interference power was reduced over a broad frequency range and the beam width was narrowed by using the diffused sensing.

Enhancement of Sound Sources Located within a Particular Area Using a Pair of Small Microphone Arrays

A method for extracting a sound signal from a particular area that is surrounded by multiple ambient noise sources is proposed. This method performs several fixed beamformings on a pair of small microphone arrays separated from each other to estimate the signal and noise power spectra. Noise suppression is achieved by applying spectrum emphasis to the output of fixed beamforming in the frequency domain, which is derived from the estimated power spectra. In experiments performed in a room with reverberation, this method succeeded in suppressing the ambient noise, giving an SNR improvement of more than 10 dB, which is better than the performance of the conventional fixed and adaptive beamforming methods using a large-aperture microphone array. We also confirmed that this method keeps its performance even if the noise source location changes continuously or abruptly.

Analytical approach to transforming filter design for sound field recording and reproduction using circular arrays with a spherical baffle

A sound field recording and reproduction method using circular arrays of microphones and loudspeakers with a spherical baffle is proposed. The spherical baffle is an acoustically rigid object on which the microphone array is mounted. The driving signals of the loudspeakers must be obtained from the signals received by the microphones. A transform filter for this signal conversion is analytically derived, which is referred to as the wave field reconstruction filter. The proposed method using a spherical baffle is compared with methods using an array of directional microphones and a microphone array mounted on a cylindrical baffle. Numerical simulations indicated that the proposed method is advantageous for sound field recording and reproduction compared with the other two methods. The results of measurement experiments in a real environment are also demonstrated.

Design of transform filter for sound field reproduction using microphone array and loudspeaker array

In this paper, we propose a novel method of sound field reproduction using a microphone array and loudspeaker array. Our objective is to obtain the driving signal of a planar or linear loudspeaker array only from the sound pressure distribution acquired by the planar or linear microphone array. In this study, we derive a formulation of the transform from the received signals of the microphone array to the driving signals of the loudspeaker array. The transform is achieved as a mean of a filter in a spatio-temporal frequency domain. Numerical simulation results are presented to compare the proposed method with the method based on the conventional least square algorithm. The reproduction accuracies were found to be almost the same, however, the filter size and amount of calculation required for the proposed method were much smaller than those for the least square algorithm based one.