ACOUSTIC SIGNAL ENHANCEMENT APPARATUS, METHOD AND PROGRAM

Abstract

Provided is an acoustic signal enhancement device, including a time-space covariance matrix estimation unit 2 configured to estimate a time-space covariance matrix Rf(n),Pf(n) corresponding to a sound source n, using a power λt,f(n) of the sound source n and an observation signal vector Xt,f composed of an observation signal xm,t,f from a microphone m;a reverberation suppression unit 3 configured to obtain a reverberation removal filter Gf(n) of the sound source n using the time-space covariance matrix Rf(n),Pf(n), and to generate a reverberation suppression signal vector Zt,f(n) corresponding to the observation signal xm,t,f for an emphasized sound of the sound source n using the reverberation removal filter Gf(n) and the observation signal vector Xt,f; anda sound source separation unit 4 configured to obtain an emphatic sound yt,f(n) of the sound source n and the power λt,f(n) of the sound source n using the reverberation suppression signal vector Zt,f(n).

Description

TECHNICAL FIELD

The present invention relates to an acoustic signal enhancement technology for separating an acoustic signal, in which a plurality of sounds and reverberations thereof collected by a plurality of microphones are mixed, into individual sounds without previous information on each of sound components, while simultaneously suppressing reverberation.

BACKGROUND ART

A conventional method 1 of the acoustic signal enhancement technology includes a reverberation suppression step of simultaneously suppressing reverberation related to all sound components without previous information on each sound component, and a sound source separation step of separating mixed sounds after the reverberation suppression into individual sounds. A configuration of the conventional method 1 is illustrated in FIG. 4.

A conventional method 2 of the acoustic signal enhancement technology includes the same processing steps as those of the conventional method 1. However, in the conventional method 2, the optimum processing can be performed by repeating the steps of feeding back the sound source separation results to the reverberation suppression step and processing each block again. A configuration of the conventional method 2 is illustrated in FIG. 5.

CITATION LIST

Non Patent Literature

• [NPL 1] Takaaki Hori, Shoko Araki, Takuya Yoshioka, Masakiyo Fujimoto, Shinji Watanabe, Takanobu Oba, Atsunori Ogawa, Kazuhiro Otsuka, Dan Mikami, Keisuke Kinoshita, Tomohiro Nakatani, Atsushi Nakamura, Junji Yamato, “Low-latency real-time meeting recognition and understanding using distant microphones and omni-directional camera”, IEEE Trans. Audio, Speech, and Language Processing, vol. 20, No. 2, pp. 499-513, 2011.
• [NPL 2] Takuya Yoshioka, Tomohiro Nakatani, Masato Miyoshi, Hiroshi G Okuno, “Blind separation and dereverberation of speech mixtures by joint optimization”, IEEE Trans. Audio, Speech, and Language Processing, vol. 19, No. 1, pp. 69-84, 2010.

SUMMARY OF INVENTION

Technical Problem

However, in the conventional method 1, since the reverberation suppression step is performed independently of processing performed in the sound source separation step to be executed subsequently, the reverberation suppression and the sound source separation are performed at the same time, whereby the optimum processing cannot be achieved.

In the conventional method 2, generally optimum processing is possible, however it is necessary to obtain a matrix of a sufficiently large size for all sound sources, all microphones and all filter coefficients and to calculate its inverse matrix for estimating the reverberation suppression again when the sound source separation results are fed back. Therefore, a large calculation cost is incurred.

An object of the present invention is to provide an acoustic signal enhancement device, a method and a program, each of which can achieve calculation costs lower than those of the conventional methods.

Solution to Problem

An acoustic signal enhancement device according to one aspect of the present invention includes:

• • a time-space covariance matrix estimation unit configured to estimate a time-space covariance matrix R_f⁽ⁿ⁾,P_f⁽ⁿ⁾ corresponding to a sound source n, using a power λ_t,f⁽ⁿ⁾ of the sound source n and an observation signal vector X_t,f composed of an observation signal x_m,t,f from a microphone m, wherein t denotes a time frame number, f denotes a frequency number, N denotes the number of sound sources, M denotes the number of microphones, n is any number from 1 to N, and m is any number from 1 to M;
  • a reverberation suppression unit configured to obtain a reverberation removal filter G_f⁽ⁿ⁾ of the sound source n using the estimated time-space covariance matrix R_f⁽ⁿ⁾,P_f⁽ⁿ⁾, and to generate a reverberation suppression signal vector Z_t,f⁽ⁿ⁾ corresponding to the observation signal x_m,t,f for an emphasized sound of the sound source n using the obtained reverberation removal filter G_f⁽ⁿ⁾ and the observation signal vector X_t,f;
  • a sound source separation step configured to obtain an emphatic sound y_t,f⁽ⁿ⁾ of the sound source n and the power λ_t,f⁽ⁿ⁾ of the sound source n using the generated reverberation suppression signal vector Z_t,f⁽ⁿ⁾; and
  • a control unit configured to control repeated processing of the time-space covariance matrix estimation unit, the reverberation suppression unit, and the sound source separation unit.

Advantageous Effects of Invention

Different from the conventional method 1, the optimum processing can be achieved by repeated processing. Since it is not necessary to consider a relationship among sound sources in the reverberation removal of the present invention, a size of a matrix necessary for optimization can be greatly reduced as compared with the conventional method 2. Therefore, the total calculation cost can be reduced.

BRIEF DESCRIPTION OF DRAWINGS

FIG. 1 is a diagram illustrating an example of a functional configuration of an acoustic signal enhancement device.

FIG. 2 is a diagram illustrating an example of a processing procedure of an acoustic signal enhancement method.

FIG. 3 is a diagram illustrating an example of a functional configuration of a computer.

FIG. 4 is a diagram illustrating a configuration of a conventional method 1.

FIG. 5 is a diagram illustrating a configuration of a conventional method 2.

DESCRIPTION OF EMBODIMENTS

Embodiments of the present invention will be described hereinafter in detail. Further, components with the same function are denoted by the same reference numerals in the diagrams, and redundant description will be omitted accordingly.

[Acoustic Signal Enhancement Device and Method]

As shown in FIG. 1, the acoustic signal enhancement device includes an initialization unit 1, a time-space covariance matrix estimation unit 2, a reverberation suppression unit 3, a sound source separation unit 4 and a control unit 5, for example.

An acoustic signal enhancement method is implemented, for example, by causing each constituent unit of the acoustic signal enhancement device to execute processing from step S1 to step S5 shown in FIG. 2 to be described below.

A symbol “−” used in the following description should be denoted immediately above a character following the symbol, but it is herein denoted immediately in front of the character due to the limitation of text notation. These symbols are written at positions where they should have been, that is, right above the characters. For example, “⁻X” in the description will be displayed in the equation as follows:

[Math. 1]

X

First, usage of symbols will be described hereinbelow.

M is the number of microphones, and m (1≤m≤M) is a microphone number. M is a positive integer equal to or greater than 2.

N is the number of sound sources, and n (1≤n≤N) is a sound source number. Note that the sound source number is represented by an upper right subscript. For example, it is represented as ⁻w⁽ⁿ⁾. N is a positive integer equal to or greater than 2.

t and τ (1≤t, τ≤T) are time frame numbers. T is the total number of time frames, and is a positive integer of 2 or more.

f (1≤f≤F) is a frequency number. The sound source is represented by an upper right subscript, and the microphone, time and frequency are represented by lower right subscripts. For example, it is represented by z_m,t,f⁽ⁿ⁾. F is a frequency corresponding to the highest frequency bin.

(⋅)^T denotes a non-conjugate transpose of a matrix or vector, and (⋅)^H denotes a conjugate transpose a matrix or vector. ⋅ indicates any matrix or vector.

Lowercase letters are scalar variables. For example, an observation signal x_m,t,f for a microphone m is a scalar variable wherein t denotes a time and f denotes a frequency.

Uppercase letters represent vectors or matrices. For example, X_t,f=[x_1,t,f, x_2,t,f, . . . , x_M,t,f]^T∈C^M×1 is an observation signal vector for all microphones wherein t denotes a time and f denotes a frequency.

C^M×N indicates the entire set of M×N-dimensional complex matrices. X∈C^M×N is a notation indicating that it is an element of the matrix. That is, it indicates X is an element of C^M×N.

λ_t,f⁽ⁿ⁾ is a power of the sound source n at the time t and the frequency f, which is a scalar.

y_t,f⁽ⁿ⁾ is an emphatic sound of the sound source n at the time t and the frequency f, which is a scalar.

G_f⁽ⁿ⁾∈C^M(L-D)×M is a reverberation removal filter for the sound source n at the frequency f. L is a filter order and is a positive integer of 2 or more. D is an expected delay and is a positive integer of 1 or more.

W_f=[Q_f⁽¹⁾, Q_f⁽²⁾, . . . , Q_f^(N)]^T∈C^M×N is a separation matrix for the frequency f, and Q_f⁽ⁿ⁾ is a separation filter for the sound source n at the frequency f.

R_f⁽ⁿ⁾∈C^{M(L-D)×M(L-D)}, P_f⁽ⁿ⁾∈C^M(L-D)×M is a time-space covariance matrix for each sound source at the frequency f.

Each constituent unit of the acoustic signal enhancement device will be described below.

<Initialization Unit 1>

When n is any number from 1 to N, the initialization unit 1 initializes a power λ_t,f⁽ⁿ⁾ and a separation matrix W_f=[Q_f⁽¹⁾, Q_f⁽²⁾, . . . , Q_f^(N)]^T∈C^M×N for each sound source n.

The initialized power λ_t,f⁽ⁿ⁾ for the sound source n is output to the time-space covariance matrix estimation unit 2. The initialized separation matrix W_f is output to the sound source separation unit 4. The initialized power λ_t,f⁽ⁿ⁾ for the sound source n may be output to the sound source separation unit 4, if needed.

The initialization unit 1 initializes the power λ_t,f⁽ⁿ⁾ and the separation matrix W_f for the sound source n. For example, the initialization unit 1 initializes the variables with a separation filter Q_f⁽ⁿ⁾ of the sound source n as an identity matrix, and with the power λ_t,f⁽ⁿ⁾ of the sound source n as a power of the observation signal x_n,t,f. The initialization unit 1 may initialize these variables by another method.

<Time-Space Covariance Matrix Estimation Unit 2>

The power λ_t,f⁽ⁿ⁾ of the sound source n, which has been initialized by the initialization unit 1 or updated by the sound source separation unit 4, and an observation signal vector X_t,f composed of the observation signal x_m,t,f from the microphone m are input to the time-space covariance matrix estimation unit 2.

When n is any number from 1 to N, the time-space covariance matrix estimation unit 2 estimates a time-space covariance matrix R_f⁽ⁿ⁾,P_f⁽ⁿ⁾ corresponding to the sound source n, using the power λ_t,f⁽ⁿ⁾ of the sound source n and the observation signal vector X_t,f composed of the observation signal x_m,t,f from the microphone m (step S2).

That is, the time-space covariance matrix estimation unit 2 estimates time-space covariance matrices R_f⁽¹⁾,P_f⁽¹⁾, . . . , R_f^(N),P_f^(N) corresponding to the sound sources 1, . . . , N, respectively. Unlike the conventional method 2, it is possible to achieve a lower calculation cost as compared to the conventional method 2 by estimating the time-space covariance matrix R_f⁽ⁿ⁾,P_f⁽ⁿ⁾ for each of sound sources 1, . . . , N.

The estimated time-space covariance matrix R_f⁽ⁿ⁾,P_f⁽ⁿ⁾ is output to the reverberation suppression unit 3.

The time-space covariance matrix estimation unit 2 estimates the time-space covariance matrix R_f⁽ⁿ⁾,P_f⁽ⁿ⁾ based on, for example, the following equation:

$\begin{array}{cc}{R}_f^{\left(n\right)}=\frac{1}{T}\sum _t\frac{{\stackrel{\_}{X}}_{t,f}{\stackrel{\_}{X}}_{t,f}^H}{{\lambda }_{t,f}^{\left(n\right)}}\in C^{M\left(L-D\right)\times m\left(L-D\right)}& \left[\mathrm{Math}.2\right]\end{array}$ $\begin{array}{cc}{P}_f^{\left(n\right)}=\frac{1}{T}\sum _t\frac{{\stackrel{\_}{X}}_{t,f}{X}_{t,f}^H}{{\lambda }_{t,f}^{\left(n\right)}}\in C^{M\left(L-D\right)\times M}-X_{t,f}\mathrm{is}-X_{t,f}={\left[X_{t-D,f}^T,\dots ,X_{t-L+1,f}^T\right]}^T.& \left[\mathrm{Math}.3\right]\end{array}$

In the first processing, the time-space covariance matrix estimation unit 2 executes the processing using the power λ_t,f⁽ⁿ⁾ of the sound source n, which has been initialized by the initialization unit 1. In the second processing and subsequent processing, the time-space covariance matrix estimation unit 2 executes the processing using the power λ_t,f⁽ⁿ⁾ of the sound source n, which has been updated by the sound source separation unit 4.

<Reverberation Suppression Unit 3>

The time-space covariance matrix R_f⁽ⁿ⁾,P_f⁽ⁿ⁾ corresponding to each sound source n, estimated by the time-space covariance matrix estimation unit 2, and the observation signal vector X_t,f composed of the observation signal x_m,t,f from the microphone m are input to the reverberation suppression unit 3.

The reverberation suppression unit 3 obtains a reverberation removal filter G_f⁽ⁿ⁾ of the sound source n using the time-space covariance matrix R_f⁽ⁿ⁾,P_f⁽ⁿ⁾, wherein n is any number from 1 to N, and generates a reverberation suppression signal vector Z_t,f⁽ⁿ⁾ corresponding to the observation signal x_m,t,f, associated with an emphatic sound of the sound source n, using the obtained reverberation removal filter G_f⁽ⁿ⁾ and the observation signal vector X_t,f (step S3).

In other words, the reverberation suppression unit 3 generates reverberation suppression signal vectors Z_t,f⁽¹⁾, . . . , Z_t,f^(N) corresponding to the observational signal x_m,t,f, associated with empathic sounds of the sound sources 1, . . . , N, respectively. Z_t,f⁽ⁿ⁾ is [z_1,t,f⁽ⁿ⁾, . . . , z_M,t,f⁽ⁿ⁾], m is any number from 1 to M, and z_m,t,f⁽ⁿ⁾ is a reverberation suppression signal corresponding to the observation signal x_m,t,f, associated with the empathic sound of the sound source n.

The generated reverberation suppression signal vector Z_t,f⁽ⁿ⁾ is output to the sound source separation unit 4.

The reverberation suppression unit 3 obtains the reverberation removal filter G_f⁽ⁿ⁾ based on, for example, the following equation:

[Math. 4]

G _f ⁽ⁿ⁾=(R_f⁽ⁿ⁾)⁻¹P_f⁽ⁿ⁾

Moreover, the reverberation suppression unit 3 obtains the reverberation suppression signal vector Z_t,f⁽ⁿ⁾ based on, for example, the following equation:

[Math. 5]

Z _t,f ⁽ⁿ⁾ =X _t,f−(G_f⁽ⁿ⁾)^HX_t,f

<Sound Source Separation Unit 4>

The reverberation suppression signal vector Z_t,f⁽ⁿ⁾, which has been generated by the reverberation suppression unit 3, is input to the sound source separation unit 4.

The sound source separation unit 4 obtains an emphatic sound y_t,f⁽ⁿ⁾ of the sound source n and the power λ_t,f⁽ⁿ⁾ of the sound source n, using the generated reverberation suppression signal vector Z_t,f⁽ⁿ⁾, wherein n is any number from 1 to N (step S4).

In particular, the sound source separation unit 4 generates the separation filter Q_f⁽ⁿ⁾ corresponding to the sound source n using the generated reverberation suppression signal vector Z_t,f⁽ⁿ⁾, wherein n is any number from 1 to N; obtains the emphatic sound y_t,f⁽ⁿ⁾ of the sound source n using the generated separation filter Q_f⁽ⁿ⁾ and the generated reverberation suppression signal vector Z_t,f⁽ⁿ⁾; and then, obtains the power λ_t,f⁽ⁿ⁾ of the sound source n, using the obtained emphatic sound y_t,f⁽ⁿ⁾.

The obtained emphatic sound y_t,f⁽ⁿ⁾ of the sound source n is output from the acoustic signal enhancement device. The obtained power λ_t,f⁽ⁿ⁾ of the sound source n is output to the time-space covariance matrix estimation unit 2.

An example of the processing in the sound source separation unit 4 will be described hereinbelow. The sound source separation unit 4 may obtain the emphatic sound y_t,f⁽ⁿ⁾ of the sound source n and the power λ_t,f⁽ⁿ⁾ of the sound source n with conventional methods other than a method to be described below.

In this example, the power λ_t,f⁽ⁿ⁾ of the sound source n, which has been initialized by the initialization unit 1, is further input to the sound source separation unit 4.

The sound source separation unit 4 repeatedly executes (1) processing of obtaining a spatial covariance matrix Σ_Z,f⁽ⁿ⁾ corresponding to the sound source n using the reverberation suppression signal vector Z_t,f⁽ⁿ⁾ and the power λ_t,f⁽ⁿ⁾ of the sound source n; (2) processing of updating the separation filter Q_f⁽ⁿ⁾ corresponding to the sound source n using the obtained spatial covariance matrix Σ_Z,f⁽ⁿ⁾, (3) processing of updating the emphatic sound y_t,f⁽ⁿ⁾ of the sound source n using the updated separation filter Q_f⁽ⁿ⁾ and the reverberation suppression signal vector Z_t,f⁽ⁿ⁾; and (4) processing of updating the power λ_t,f⁽ⁿ⁾ of the sound source n using the updated emphatic sound y_t,f⁽ⁿ⁾, thereby finally obtaining the emphatic sound y_t,f⁽ⁿ⁾ of the sound source n, wherein n is any number from 1 to N.

In other words, the sound source separation unit 4 repeatedly executes (1) processing of obtaining spatial covariance matrices Σ_Z,f⁽¹⁾, . . . , Σ_Z,f^(N) corresponding to the sound sources 1, . . . , N, respectively, using the reverberation suppression signal vectors Z_Z,f⁽¹⁾, . . . , Z_Z,f^(N) and the power λ_t,f⁽¹⁾, . . . , λ_t,f^(N) of the sound sources 1 to N; (2) processing of updating the separation filters Q_f⁽¹⁾, . . . , Q_f^(N) corresponding to the sound sources 1, . . . , N, respectively, using the obtained spatial covariance matrices Σ_Z,f⁽¹⁾, . . . , Σ_Z,f^(N); (3) processing of updating the emphatic sounds y_t,f⁽¹⁾, . . . , y_t,f^(N) of the sound sources 1, . . . , N, respectively, using the updated separation filters Q_f⁽¹⁾, . . . , Qf^(N) and the reverberation suppression signal vectors Z_t,f⁽¹⁾, . . . , Z_t,f^(N); and (4) processing of updating the powers λ_t,f⁽¹⁾, . . . , λ_t,f^(N) of the sound sources 1, . . . , N, respectively, using the updated emphatic sounds y_t,f⁽¹⁾, . . . , y_t,f^(N), thereby finally obtaining the emphatic sounds y_t,f⁽¹⁾, . . . , y_t,f^(N) of the sound sources 1, . . . , N, respectively.

The processing (1) to (4) need not be repeated. That is, the processing (1) to (4) may be performed only once in a single processing of step S4.

The finally obtained emphatic sound y_t,f⁽ⁿ⁾ of the sound source n is output from the acoustic signal enhancement device. The finally updated power λ_t,f⁽ⁿ⁾ of the sound source n is output to the time-space covariance matrix estimation unit 2.

The sound source separation unit 4 obtains the spatial covariance matrix Σ_Z,f⁽ⁿ⁾ based on, for example, the following equation:

$\begin{array}{cc}{\sum }_{Z,f}^{\left(n\right)}=\frac{1}{T}{\sum }_t{Z_{t,f}^{\left(n\right)}\left(Z_{t,f}^{\left(n\right)}\right)}^H/{\lambda }_{t,f}^{\left(n\right)}& \left[\mathrm{Math}.6\right]\end{array}$

The sound source separation unit 4 updates the separation filter Q_f⁽ⁿ⁾ based on, for example, the following Equation (1) and (2). More specifically, it updates the separation filter Q_f⁽ⁿ⁾ by plugging Q_f⁽ⁿ⁾ obtained by Equation (1) into the right side of Equation (2) and calculating Q_f⁽ⁿ⁾ defined by Equation (2).

[Math. 7]

Q _f ⁽ⁿ⁾=(W_f^HΣ_Z,f⁽ⁿ⁾)⁻¹e_n  (1)

[Math. 8]

Q _f ⁽ⁿ⁾=((Q_f⁽ⁿ⁾)^HΣ_Z,f⁽ⁿ⁾Q_f⁽ⁿ⁾)^−1/2Q_f⁽ⁿ⁾  (2)

n is any number from 1 to N, and e_n is an N-dimensional vector wherein an n-th element is 1 and other elements are 0.

The sound source separation unit 4 updates the emphatic sound y_t,f⁽ⁿ⁾ of the sound source n based on, for example, the following equation:

[Math. 9]

y _t,f ⁽ⁿ⁾=(Q_f⁽ⁿ⁾)^HZ_t,f⁽ⁿ⁾

The sound source separation unit 4 updates the power λ_t,f⁽ⁿ⁾ of the sound source n based on, for example, the following equation:

$\begin{array}{cc}{\lambda }_t^{\left(n\right)}=\frac{1}{F}\sum _{f=0}^{F-1}{❘y_{t,f}^{\left(n\right)}❘}^2& \left[\mathrm{Math}.10\right]\end{array}$

By feeding back the sound source separation results to the processing of the reverberation suppression unit 3 and repeating all the processing, the optimum processing can be achieved unlike the conventional method 1. Further, since it is not necessary to consider a relationship between sound sources for each sound source by estimating the time-space covariance matrix R_f⁽ⁿ⁾,P_f⁽ⁿ⁾ for each sound source, the matrix required for optimization can has a greatly reduced size as compared to the conventional method 2. Therefore, the total calculation cost can be reduced.

Although the processing of the sound source separation unit 4 usually requires a large number of repetitions until convergence, the calculation cost of each repeated processing is smaller than that of the processing executed by the reverberation suppression unit 3. Therefore, convergence can be obtained at a smaller calculation cost by repeating the processing of the sound source separation unit 4 several times in each of the repeated processing. Thus, more flexible control can be achieved by setting the processing of the reverberation suppression unit 3 separately from the processing of the sound source separation unit 4.

<Control Unit 5>

A control unit 5 controls the repeated processing of the time-space covariance matrix estimation unit 2, the reverberation suppression unit 3, and the sound source separation unit 4 (step S5).

For example, the control unit 5 performs the repeated processing until a predetermined termination condition is satisfied. An example of the predetermined termination condition is that predetermined variables converge, such as the emphatic sound y_t,f⁽ⁿ⁾ of the sound source n. Another example of the predetermined termination condition is that the number of times of repeated processing reaches a predetermined number of times.

[Modification]

While an embodiment of the present invention has been described above, specific configurations are not limited to the embodiment, and it will be appreciated that the present invention also encompasses modifications or alterations without departing from the spirit and the scope of the invention.

The various processing explained in the embodiment may not only be executed in chronological order according to the described sequences but may also be executed in parallel or individually in accordance with processing capability of a device to be used to execute the processing or as necessary.

For example, data may be directly exchanged between the constituent units of the acoustic signal enhancement device or may be exchanged via a storage unit (not illustrated).

[Program and Recording Medium]

The processing of each unit of each device may be implemented by a computer, and in this case, the processing details of the functions that each device should have are described by a program. The various types of processing functions of each device are implemented on a computer, by causing this program to be loaded onto a storage unit 1020 of the computer 1000 shown in FIG. 3, and operating, for example, a computational processing unit 1010, an input unit 1030 and an output unit 1040.

The program describing the processing contents can be recorded on a computer-readable recording medium. The computer-readable recording medium is, for example, a non-transitory recording medium, and specifically, for example, a magnetic recording device or an optical disc.

The program is distributed, for example, by sales, transfer, or rent of a portable recording medium such as a DVD or a CD-ROM on which the program is recorded. In addition, the distribution of the program may be performed by storing the program in advance in a storage device of a server computer and transferring the program from the server computer to another computer via a network.

A computer executing such a program is configured to, for example, first, temporarily store a program recorded on a portable recording medium or a program transferred from a server computer in an auxiliary recording unit 1050 which is its own non-temporary storage device. When executing the processing, the computer reads the program stored in the auxiliary recording unit 1050 which is its own non-temporary storage device into the storage unit 1020, and executes the processing according to the read program. As another embodiment of the program, the computer may directly read the program from the portable recording medium into the storage unit 1020 and execute processing according to the program. Each time the program is transferred from the server computer to the computer, the processing according to the received program may be executed sequentially. In addition, the processing may be executed by means of a so-called application service provider (ASP) service which does not transfer a program from the server computer to the computer and implements processing functions only by execution instructions and acquisition of the results. It is assumed that the program in this embodiment includes data which is information to be provided for processing by an electronic computer and which is equivalent to a program (e.g. data that is not a direct command to the computer but has the property of defining the processing of the computer).

In this aspect, the device is configured by executing a predetermined program on a computer, but at least a part of the processing may be implemented by hardware.

In addition, changes, alterations or modifications can be made as appropriate without departing from the gist of the present invention.

Claims

1. An acoustic signal enhancement device, comprising: processing circuitry configured to implement: a time-space covariance matrix estimation unit configured to estimate a time-space covariance matrix Rf(n),Pf(n) corresponding to a sound source n, using a power λt,f(n) of the sound source n and an observation signal vector Xt,f composed of an observation signal xm,t,f from a microphone m, wherein t denotes a time frame number, f denotes a frequency number, N denotes the number of sound sources, M denotes the number of microphones, n is any number from 1 to N, and m is any number from 1 to M;a reverberation suppression unit configured to obtain a reverberation removal filter Gf(n) of the sound source n using the estimated time-space covariance matrix Rf(n),Pf(n), and to generate a reverberation suppression signal vector Zt,f(n) corresponding to the observation signal xm,t,f for an emphasized sound of the sound source n using the obtained reverberation removal filter Gf(n) and the observation signal vector Xt,f;a sound source separation unit configured to obtain an emphatic sound yt,f(n) of the sound source n and the power λt,f(n) of the sound source n using the generated reverberation suppression signal vector Zt,f(n); anda control unit configured to control repeated processing of the time-space covariance matrix estimation unit, the reverberation suppression unit, and the sound source separation unit.

2. The acoustic signal enhancement device according to claim 1, wherein the sound source separation unit is configured to repeatedly execute (1) processing of obtaining a spatial covariance matrix ΣZ,f(n) corresponding to the sound source n using the generated reverberation suppression signal vector Zt,f(n) and the power λt,f(n) of the sound source n; (2) processing of updating a separation filter Qf(n) corresponding to the sound source n using the obtained spatial covariance matrix ΣZ,f(n); (3) processing of updating the emphatic sound yt,f(n) of the sound source n using the updated separation filter Qf(n) and the generated reverberation suppression signal vector Zt,f(n); and (4) processing of updating the power λt,f(n) of the sound source n using the updated emphatic sound yt,f(n), thereby finally obtaining the emphatic sound yt,f(n) of the sound source n.

3. An acoustic signal enhancement method, comprising: time-space covariance matrix estimation by a time-space covariance matrix estimation unit, by estimating a time-space covariance matrix Rf(n),Pf(n)) corresponding to a sound source n, using a power λt,f(n) of the sound source n and an observation signal vector Xt,f composed of an observation signal xm,t,f from a microphone m, wherein t denotes a time frame number, f denotes a frequency number, N denotes the number of sound sources, M denotes the number of microphones, n is any number from 1 to N, and m is any number from 1 to M;reverberation suppression by a reverberation suppression unit, by obtaining a reverberation removal filter Gf(n) of the sound source n using the estimated time-space covariance matrix Rf(n),Pf(n), and generating a reverberation suppression signal vector Zt,f(n) corresponding to the observation signal xm,t,f for an emphasized sound of the sound source n using the obtained reverberation removal filter Gf(n) and the observation signal vector Xt,f;sound source separation by a sound source separation unit, by obtaining an emphatic sound yt,f(n) of the sound source n and the power λt,f(n) of the sound source n using the generated reverberation suppression signal vector Zt,f(n); andby a control unit, controlling repeated processing of the time-space covariance matrix estimation, the reverberation suppression, and the sound source separation.

4. A non-transitory computer readable medium that stores a program for causing a computer to perform as each step of the acoustic signal enhancement method according to claim 3.