JP4586577B2 - Disturbance component suppression device, computer program, and speech recognition system - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a disturbance component suppressing device which improves speech recognizing performance in actual environment wherein there is an influence of disturbance such as additive noise and reverberation is present and can suppress components of disturbance in a short time. <P>SOLUTION: A disturbance component suppression section 114 includes a disturbance probability distribution estimation portion 200 which receives feature quantities 124 extracted frames of a designated time length divided by designated periods as to an observation signal obtained by observing a target speech in environment wherein additive noise and multiplicative distortion are generated and generates parameters 206 representing disturbance by frames one after another by using a particle filter having a plurality of particles, and a parameter generation portion 202 and a screen speech estimation portion 204 which calculate estimated feature quantities 126 of a target speech by the frames by using the feature quantities 124 of the observation signal, estimated parameters 206 of the disturbance, and a GMM 130. <P>COPYRIGHT: (C)2006,JPO&amp;NCIPI

Description

  The present invention relates to speech recognition technology in a real environment where disturbances that affect speech occur, and in particular, speech in an environment where non-stationary additive noise and multiplicative distortion such as reverberation occur. The present invention relates to a disturbance component suppressing device and a speech recognition system using the same to improve recognition accuracy.

  Speech recognition technology has been studied as a technology for realizing a human machine interface that is easy and natural for humans. In recent years, speech recognition at a high recognition rate has been realized by a large-scale speech / text database and statistical stochastic speech recognition techniques. Nowadays, applied technology is being developed to realize speech recognition at high speed and with a high recognition rate in a real environment where a human is in contact with a machine.

  One of the major differences between the actual environment and the laboratory environment is the presence of noise. Noise occurs constantly and irregularly at a volume that cannot be ignored. In addition, in the real environment, multiplicative distortion occurs in the voice depending on the spatial transfer characteristics of the voice in the environment or due to reverberation or the like. Such disturbance disturbs voice recognition. Improvement of speech recognition performance in an environment where these disturbances occur is a problem that should be solved as soon as possible in developing an application technology for speech recognition.

  One technique for improving speech recognition performance in a noisy environment is a technique for estimating and suppressing noise at the pre-processing stage of speech recognition. Non-Patent Document 1 described later discloses a spectral subtraction method that is a general method of noise suppression. In this method, it is assumed that the noise amplitude spectrum observed in the interval before the utterance is the same as the noise amplitude spectrum in the utterance interval, and from the amplitude spectrum of the speech signal obtained from the utterance, immediately before the utterance. Noise is suppressed by subtracting the amplitude spectrum of the observed noise.

  There is also a technique for sequentially estimating and suppressing noise in the preprocessing stage of speech recognition. Non-Patent Document 2 discloses a technique of sequentially obtaining a maximum likelihood estimation value of noise by applying a sequential EM (Expectation Maximization) algorithm. In the technique of sequentially estimating noise using the sequential EM algorithm, it is possible to estimate and suppress noise with high accuracy while coping with temporal fluctuation of noise.

  A method of sequentially obtaining an estimated value of noise using a Kalman filter disclosed in Non-Patent Document 3 and Non-Patent Document 4 is also generally used. In this method, noise is sequentially estimated and suppressed by alternately performing first-term prediction and filtering.

  As a technique for improving speech recognition performance in a noisy environment, there is a technique for performing adaptive speech recognition using a probability model that takes noise into account. For example, in Patent Document 1 described later, noise parameters are estimated using a sequential estimation method called a particle filter, and temporal growth of a hidden state constituting a Hidden Markov Model (HMM) is performed. A speech recognition system that performs speech recognition based on the hidden Markov model is disclosed.

  As a technique for improving speech recognition performance in an environment where multiplicative distortion occurs, there is a technique for removing multiplicative distortion using a Cepstrum Mean Subtraction (CMS). This technique can remove multiplicative distortions having a transfer characteristic with an impulse response length shorter than the analysis window length, such as distortion due to the characteristics of a recording microphone.

Non-Patent Document 5 discloses a technique for performing speech recognition under reverberation by regarding reflected sound as additive noise. In this technique, speech observed under reverberation (hereinafter referred to as “reverberation speech”) is expressed by first-order linear prediction. Here, the vector having a linear Mel spectrum of the voice and reverberant sound component and S _t ^Lin, and X _S, and _t ^Lin at time t, the transfer characteristic of the audio at each Mel frequency domain, i.e. the multiplicative distortion linear Let H ^Lin be a matrix having a mel spectrum as a diagonal component. A matrix having a linear prediction coefficient of reverberation in each mel frequency region as a diagonal component is A ^Lin . In this technique, a reverberant speech vector X _{S, t} ^Lin is expressed by the following recursive formula.

X _{S, t} ^Lin = H ^Lin S _t ^Lin + A ^Lin X _{S, t-1} ^Lin
In this technique, the elements of the matrix H ^Lin , that is, the linear mel spectrum of multiplicative distortion, and the elements of the matrix A ^Lin , that is, the linear prediction coefficient of reverberation are regarded as time-fixed parameters, and these parameters are estimated by the EM algorithm. To do. Since the impulse response length distortion longer than the analysis window length is expressed by the above recursive formula, the influence of reflected sound and the like can be modeled.

JP 2002-251198 A S. F. Bol: “Suppression of acoustic noise in speech using spectral subtraction”, IEEE Trans. ASSP, Vol. 27, no. 2, 113-120, 1979 (S.F. Boll: “Suppression of Acoustic Noise in Speech Using Spectral Subtraction,” IEEE Trans. ASSP, Vol. 27, No. 2, pp. 113-120, 1979) M.M. Affifi, O. Shioan: “Sequential estimation with optimal forgetting for robust speech recognition”, IEEE Trans. SAP, Vol. 12, no. 1, 19-26, 2004 (M. Afify, O. Siohan: “Sequential Estimation with Optimal Forgetting for Robust Speech Recognition,” IEEE Trans. SAP, Vol. 12, No. 1, pp. 19-26, 2004. ) Takashi Arimoto: “Kalman Filter”, industrial books Supervised by Michio Nakano, Kiyoshi Nishiyama: “Kalman filter solved on a personal computer”, Maruzen Tetsugu Takiguchi et al .: “Recognition of reverberant speech using first-order linear prediction”, ICASSP '04, pages 869-872, May 2004 (T. Takiguchi et al: “Reverberant speech recognition using first-order linear prediction,” ICASSP'04, pp. 896-872, May 2004)

  In the real environment, most of the noise is non-stationary noise. That is, the acoustic characteristics of noise vary with time. As in the spectral subtraction method described in Non-Patent Document 1, a technique that estimates and suppresses noise under the premise that noise is stationary cannot cope with time fluctuations of noise and is highly accurate. Noise cannot be suppressed.

  The method using the sequential EM algorithm described in Non-Patent Document 2 performs iterative calculation until the value converges to the local optimum value of the likelihood function. Therefore, an enormous amount of calculation is required every time the noise fluctuates, and the calculation takes time. Therefore, it is difficult to estimate and suppress noise in real time by this method.

  The estimation method using the Kalman filter disclosed in Non-Patent Document 3 and Non-Patent Document 4 performs successive estimation by alternately performing one-time prediction and filtering. Therefore, it does not require an iterative calculation like the sequential EM algorithm. However, the method using the Kalman filter estimates the probability distribution assuming that the posterior probability distribution of noise is a single normal distribution. When the true probability distribution is a mixed distribution, it is approximated by a single normal distribution. Therefore, the accuracy is deteriorated.

  In a technique for performing speech recognition using a model that takes noise into consideration, as in the speech recognition system described in Patent Document 1, matching is performed between a speech on which noise is superimposed and a probability model. For this reason, for example, preprocessing that should be performed based on noise-free speech such as acoustic model adaptation cannot be executed.

In the technique described in Non-Patent Document 5, the effect of reflected sound is modeled by the above recursive formula. However, reverberation is generally a phenomenon that occurs when sound is observed or recorded at a point away from a sound source. In an environment where the distance between the sound source and the observation point is large, not only the reflected sound but also the noise generated in the environment surrounding the sound source and the observation point cannot be ignored. In the technique described in Non-Patent Document 5, this point is not taken into consideration. In the technique described in Non-Patent Document 5, an element of the matrix H ^Lin , that is, a linear mel spectrum of multiplicative distortion, and an element of the matrix A ^Lin , that is, a linear prediction coefficient of reverberation are regarded as time-fixed parameters. However, in a real environment, for example, a sound source and an object that reflects sound around the sound source may move. Under such circumstances, both the multiplicative distortion parameter and the linear prediction coefficient of reverberation change with time. For this reason, the technique described in Non-Patent Document 5 cannot cope with time fluctuations of reverberation and cannot cope with the influence of disturbance with high accuracy.

  Therefore, an object of the present invention is to provide a disturbance component suppression device that can improve speech recognition performance in an environment in which multiplicative distortion such as non-stationary noise and reverberation occurs, and can suppress disturbance components in a short time. Is to provide.

  A disturbance component suppressing device according to a first aspect of the present invention is a device that suppresses disturbance components of an observation signal obtained by observing a target speech in an environment where additive noise and multiplicative distortion occur. This apparatus receives a feature amount extracted from a frame of a predetermined time length that is framed every predetermined period for an observation signal, and uses a particle filter having a plurality of particles to estimate a probability distribution parameter representing a disturbance. Is calculated for each frame using disturbance parameter estimation means for sequentially generating a frame for each frame, a feature amount of an observation signal, an estimation parameter, and a predetermined acoustic model related to the target speech. And target speech estimation means.

  Preferably, the disturbance parameter estimation means sets an initial distribution of the disturbance, and an initial parameter setting means for setting an initial parameter of the probability distribution representing the disturbance in each of the plurality of particles with a probability according to the initial distribution. And an estimated parameter of the preceding first frame in each particle corresponding to the second frame following the first frame using an extended Kalman filter based on the acoustic model and the feature amount of the observation signal Updating means for updating the weights, and weight calculating means for calculating the weight of each of the plurality of particles in the second frame.

  More preferably, the initial parameter setting means estimates the initial distribution of additive noise based on the feature quantity of the observed signal, and the probability distribution of the additive noise in each of the plurality of particles with a probability according to the initial distribution. Means for sampling each of the initial parameters, and means for setting the value of the initial parameter of the multiplicative distortion probability distribution in each of the plurality of particles to a predetermined value.

  More preferably, the disturbance parameter estimation means further corresponds to the estimation parameter corresponding to the first frame in each of the plurality of particles based on the parameter resampled by the resampling means. A re-updating means for re-upding to a thing, an estimation parameter re-updated by the re-updating means in each of a plurality of particles, and an estimated parameter re-sampled by the re-sampling means And selecting means for selecting as the estimation parameter of the second frame.

  Preferably, the target speech estimation unit includes an observation signal model synthesis unit for synthesizing a probability model of the observation signal corresponding to the frame based on the feature amount of the observation signal, the estimation parameter, and the acoustic model, and the observation signal Based on the feature amount, the estimation parameter, the acoustic model, and the observed signal probability model, estimated feature amount calculation means for calculating the estimated feature amount of the target speech for each frame.

  More preferably, the observation signal model combining means includes parameter estimation means for estimating the parameters of the probability model of the observation signal for the particle based on the estimation parameter and the acoustic model for each of the plurality of particles. Including.

  The estimated feature amount calculating means is a means for calculating, for each frame, an estimation parameter of the target speech of each of the plurality of particles based on the feature amount of the observation signal, the acoustic model, the estimation parameter, and the probability model of the observation signal. And means for calculating an estimated feature amount of the target speech in the frame based on an estimation parameter of the target speech in each of the plurality of particles.

  When executed by a computer, the computer program according to the second aspect of the present invention causes the computer to operate as any of the disturbance component suppression devices according to the first aspect of the present invention.

  The speech recognition system according to the third aspect of the present invention receives the estimated feature amount of the target speech calculated by any one of the disturbance component suppressing device and the disturbance component suppressing device according to the first aspect of the present invention, Speech recognition means for performing speech recognition related to the target speech using a predetermined acoustic model related to the target speech and a predetermined language model related to the recognition target language;

Hereinafter, an embodiment of the present invention will be described with reference to the drawings. In the drawings used for the following description, the same parts are denoted by the same reference numerals. Their names and functions are also the same. Therefore, description thereof will not be repeated. The symbol “^” or the like used in the text of the following description should be described immediately above the character immediately after it, but it is described immediately before the character due to restrictions on text notation. In the formula, these symbols are written in their original positions. Also, in the text of the following description, vectors or matrices are described in the form of ordinary text with “vector”, “matrix”, etc. immediately preceding them, such as “vector X _t ”, “matrix Σ _W ”, etc. However, it is written in bold in the formula.

[Constitution]
FIG. 1 shows the overall configuration of the speech recognition system 100 according to the present embodiment. With reference to FIG. 1, the speech recognition system 100 collects a sound 122 generated by the sound source 102, and extracts a feature amount used for recognition from the collected sound, and a preprocessing unit 104. A pre-processing acoustic model unit 106 for preparing a probability model (acoustic model) connected and representing a relationship between speech and phonemes, and a probability model (language model) representing a word connection probability in a recognition target language A language model unit 108 for preparation, a search unit 110 for searching for a word or the like corresponding to the feature amount output from the preprocessing unit 104 using the language model of the language model unit 108, and a search unit 110 And a recognition acoustic model unit 109 for preparing an acoustic model that is connected and used for the search by the search unit 110.

The sound source 102 includes a speaker 116 that utters a sound (target sound) 120 to be recognized, and a disturbance factor 118 that affects sound transmission around the speaker 116. The sound 122 that reaches the pre-processing unit 104 is not the target sound 120 generated by the utterance of the speaker 116 but a sound that has changed due to the influence of the disturbance factor 118. In the present specification, the target voice 120 without noise generated by the speech of the speaker 116 is referred to as “clean voice”. The sound recorded by the preprocessing unit 104, that is, the sound 122 that reaches the preprocessing unit 104 in a state changed by the influence of the disturbance factor 118 is referred to as “observation sound”.

  The preprocessing acoustic model unit 106 prepares and holds an acoustic model composed of a Gaussian Mixture Model (GMM) for the clean speech 120. The preprocessing acoustic model unit 106 performs learning processing on the GMM using the learning data storage unit 132 for storing a large amount of learning data prepared in advance and the learning data stored in the learning data storage unit 132. Model learning unit 134 and a GMM storage unit 136 for storing GMM 130 obtained by learning by model learning unit 134.

  FIG. 2 schematically shows the concept of the GMM 130. Referring to FIG. 2, the GMM 130 is a probability model in which the value of the time series signal is modeled by one stationary signal source (state). In this GMM 130, an output probability is defined. Specifically, in the GMM 130, a value that may be output as the clean sound 120 at time t and a probability that the value is output are defined. In the GMM 130, the output probability is expressed by a mixed normal distribution. For example, the GMM 130 has a mixed normal distribution composed of single normal distributions 148A, 148B,.

  Referring again to FIG. 1, the preprocessing unit 104 records the observation sound 122 and performs predetermined signal processing on the obtained observation signal, whereby a predetermined feature vector (hereinafter simply referred to as “feature amount”) regarding the observation signal. The measurement unit 112 for extracting 124 and the disturbance component suppression unit 114 for suppressing the disturbance component included in the feature value 124 extracted by the measurement unit 112 using the GMM 130 are included.

  Specifically, the measurement unit 112 analyzes the logarithmic mel filter bank for each frame having a time length of 10 milliseconds, and outputs a vector having the obtained log mel spectrum as an element as the feature quantity 124.

  The disturbance component suppression unit 114 uses the GMM 130 to estimate the feature amount of the clean speech 120 from the feature amount 124 of the observation signal. Then, the feature amount 126 obtained by the estimation is output to the search unit 110. In this specification, the voice represented by the feature quantity 126 of the estimated clean voice is referred to as “estimated clean voice”.

FIG. 3 schematically shows a signal model of the disturbance factor 118. Referring to FIG. 3, clean speech 120 is subjected to multiplicative distortion depending on the spatial transfer characteristics from speaker 116 to measurement unit 112 shown in FIG. 1, and thus reaches measurement unit 112 from speaker 116. The sound 500 is different from the clean sound 120. Here, the time t frame (hereinafter, simply referred to as "the t frame".) The vector with the linear Mel spectra of clean speech 120 elements and S _t ^Lin in, multiplicative linear mel spectrum diagonal components of the strain Let H _t ^Lin be the matrix of Assuming that a vector having the linear mel spectrum of the sound 500 reaching the measuring unit 112 as an element is X _{S, t} ^Lin (D), X _{S, t} ^Lin (D) is generally expressed by the following expression. That is,
^{_{X S, t Lin (D)}} = H t Lin S t Lin

However, the observation sound 122 is affected by reverberation. That is, not only the sound 500 that reaches directly, but also the reflected sound 502 that is reflected by the surrounding wall surface and reaches the measuring unit 112. In the present embodiment, the reflected sound is regarded as additive noise. Assuming that a vector having the linear mel spectrum of the reflected sound 502 as an element is X _{S, t} ^Lin (R), and a sound reaching the measuring unit 112 due to the influence of reverberation is X _{S, t} ^Lin , X _{S, t} ^Lin Is expressed by the following equation. That is,
X _{S, t} ^Lin = X _{S, t} ^Lin (D) + X _{S, t} ^Lin (R)

Both the direct sound 500 and the reflected sound 502 are sounds emitted by the speaker 116, but the reflected sound 502 arrives at the measuring unit 112 with a delay from the direct sound 500 due to a difference in the propagation path. According to Non-Patent Document 5, when a matrix with linear prediction coefficients of the reverberation in the Mel frequency band diagonal and matrix ^{_{A t Lin, X S, t}} Lin (R) is represented by the following formula . That is,
X _{S, t} ^Lin (R) = A _t ^Lin X _{S, t-1} ^Lin

Further, in the actual environment, noise 504 is generated around the speaker 116 and the measurement unit 112 and reaches the measurement unit 112. Here, a vector having the linear mel spectrum of the noise 504 as an element is N _t ^Lin, and a vector having the linear mel spectrum of the observation sound 122 as an element is X _{S + N, t} ^Lin . X _{S + N, t} ^Lin can be modeled by the following signal model. That is,
X _{S + N, t} ^Lin = X _{S, t} ^Lin + N _t ^Lin = H _t ^Lin _St ^Lin + N _t ^Lin + A ^Lin X _{S, t-1} ^Lin
Since the reflected sound cannot be observed, the reflected sound vector X _{S, t-1} ^Lin is approximated as follows in this equation. That is,
X _{S, t-1} ^Lin = X _{S + N, t-1} ^Lin −N _t-1 ^Lin

A feature quantity 124 of the observation signal in the t-th frame, that is, a vector having a log mel spectrum obtained from the observation sound 122 as an element is defined as a feature quantity vector _Xt . Note that the feature vector _Xt is a vector obtained by converting each element of the vector X _{S, t} ^Lin into a log mel spectrum region. The feature vector _Xt is a known parameter obtained by measurement. Feature vector X _t is a vector vector S _t is changed by the influence of disturbance with logarithmic Mel spectra of clean speech 120 to the element. The vector _St is an unknown vector. The disturbance includes an influence due to multiplicative distortion, reverberation, and additive noise. Here, a matrix having a logarithmic mel spectrum of multiplicative distortion as a diagonal component is denoted by H _t, and a vector having a log mel spectrum of additive noise as an element is denoted by N _t . In addition, disturbances include effects due to reverberation. Further, a matrix having the logarithm of the linear prediction coefficient in the t-th frame as a diagonal component is a matrix A _t , and a vector obtained by logarithmizing each element of the reflected sound vector X _{S, t−1} ^Lin is X _{S, t−1} . .

The above-described vectors X _t , S _t , N _t , and X _{S, t−1} have the same number of dimensions. Further, the number of rows matrix H _t and A _t and the number of columns are the same. The processing described below is performed for each element of the vector and matrix. However, in the following description, each element is not particularly distinguished for the sake of simplicity.

FIG. 4 shows a state space model 160 expressing the observation process of the observation signal and the noise state change process. Referring to FIG. 4, in the state space model 160, it is assumed that the output process of the clean speech 120 can be modeled by GMM. That is, the vector S _t is a component of the clean speech 120 in the t frame is assumed to be output in accordance with an element distribution within GMM130.

In the GMM 130, let the element distribution corresponding to the t-th frame be k _t . The element distribution k _t is a single normal distribution with the mean μ _{S, kt} and the variance Σ _{S, kt} . A vector of parameters output from the element distribution k _t is a vector S _{kt, t} . Hereinafter, the parameter vector S _{kt, t} output from the GMM 130 is referred to as an “output parameter (of the GMM 130)”. A feature vector S _t of clean speech 120, the error is present between the output parameter vector S _{kt, t.} In addition, there is an error in converting X _{S + N, t} ^Lin to the log mel spectrum region. These errors are also a vector, are collectively vector of these errors, a vector V _t. A matrix representing the disturbance due to the disturbance factor 118 is Λ _t = (N _t , H _t , A _t ). The observation signal feature quantity vector X _t (124) is observed by converting the above X _{S + N, t} ^Lin into a log mel spectrum region, thereby generating an output parameter vector S _{kt, t} and an error vector V _t of the GMM 130. , And vectors X _t , S _t , N _t , and X _{S, t−1} , and matrices H _t and Λ _t are expressed by the following equation (1).

なお、式（１）でＩは単位行列を表す。また行列の対数、行列の指数演算はそれぞれ、行列の各要素について対数をとり、又は指数計算をし、その結果を成分とする行列を表すものとする。誤差ベクトルＶ_tは、次の式（２）のように平均が０で分散がΣ_S,ktの単一正規分布で表現される確率分布にしたがう値を要素に持つものとする。

In Equation (1), I represents a unit matrix. In addition, the logarithm of the matrix and the exponent operation of the matrix respectively represent a matrix having a logarithm or exponent calculation for each element of the matrix and having the result as a component. The error vector V _t has a value according to a probability distribution represented by a single normal distribution having an average of 0 and a variance of Σ _{S, kt} as shown in the following equation (2).

ただしこの式においてΣ_S,ktはＧＭＭ１３０内のある要素分布ｋ_tより得られるパラメータの共分散行列を表し、記号「〜」は左辺の値が右辺に示される確率分布にしたがうことを示す。すなわち、左辺の値が右辺に示す確率分布にしたがったサンプリングにより推定できることを示す。また、この式において、「Ｎ（μ，Σ）」は、平均値ベクトルμ、分散Σの単一正規分布を表す。

However sigma _{S, kt} in this equation represents the covariance matrix of the parameters obtained from the element distribution k _t with the GMM130, the symbol "~" indicates that according to the probability distribution values of the left side is shown in the right side. That is, the value on the left side can be estimated by sampling according to the probability distribution shown on the right side. In this equation, “N (μ, Σ)” represents a single normal distribution with an average value vector μ and a variance Σ.

In the state space model 160, it is assumed that a matrix Λ _t representing a disturbance changes according to a random walk process. That is, it is assumed that an error occurs between the matrix Λ _t−1 representing the disturbance at the t−1 frame and the matrix Λ _t representing the disturbance at time t. Vector N _t, the matrix H _t, and the error respectively for the matrix A _t, the vector W _N, the matrix W _H, and a matrix W _A, the matrix W _t = (W _Nt they collectively representing an error, W _Ht, W _At ). The time variation of the matrix Λ _t representing the disturbance is expressed by the following equation (3).

この式において、誤差行列Ｗ_t＝（Ｗ_Nt，Ｗ_Ht，Ｗ_At）を構成する、ベクトルＷ_Nt、行列Ｗ_Ht、及び行列Ｗ_Atはそれぞれ予測誤差であり、それぞれ平均０、共分散行列がΣ_WN、Σ_WH、及びΣ_WAの単一正規分布で表現される確率分布にしたがう白色性ガウス雑音であるものとする。誤差を表す行列Ｗ_tは、次の式（４）のように単一正規分布にしたがう。

In this formula, erroneous letters s column _{W t = (W Nt, W} Ht, W At) constituting the vector W _Nt, matrix W _Ht, and the matrix W _At are each prediction error, mean respectively 0, covariance It is assumed that the matrix is white Gaussian noise according to a probability distribution expressed by a single normal distribution of Σ _WN , Σ _WH , and Σ _WA . Matrix W _t representing the error, according to a single normal distribution as the following equation (4).

ただし、式（４）においてΣ_Wは、誤差を表す行列Ｗ_tの共分散行列を表す。

In Equation (4), Σ _W represents a covariance matrix of a matrix W _t representing an error.

  The disturbance component suppression unit 114 shown in FIG. 1 sequentially estimates the feature vector of clean speech for each frame using the state space model 160 expressed by the above equations (1) to (4).

FIG. 5 is a block diagram showing the configuration of the disturbance component suppressing unit 114. Referring to FIG. 5, disturbance component suppressing section 114 receives feature quantity X _t (124) of the observed signal, and uses GMM 130 to generate a probability distribution (hereinafter, “matrix Λ _t) representing the disturbance in state space model 160. A disturbance probability distribution estimation unit 200 for estimating the disturbance probability distribution, a disturbance probability distribution estimated by the disturbance probability distribution estimation unit 200, and an average vector of the probability model of the observation signal and the covariance matrix from the GMM 130 And a clean speech estimation unit 204 for calculating a feature quantity 126 of the estimated clean speech using the disturbance probability distribution, the average vector and covariance matrix of the observed signal, and the GMM 130, including.

The disturbance probability distribution estimation unit 200 has a function of sequentially estimating the disturbance probability distribution for each frame and outputting a parameter 206 representing the disturbance probability distribution (hereinafter simply referred to as “estimated disturbance distribution 206”) . Here, the matrix lambda ₀ which represents the disturbance, ..., series sequence consisting lambda _t matrix _{Λ 0: t = {Λ 0} , ..., Λ t} and. The posterior probability distribution p (Λ _{0: t} | X _{0: t} ) of the sequence Λ _{0: t} is expressed as the following equation (5) using a first-order Markov chain.

式（５）のｐ（Λ_t｜Λ_t-1）は、単一正規分布を用いて次の式（６）のようにモデル化される。

P (Λ _t | Λ _t-1 ) in the equation (5) is modeled as the following equation (6) using a single normal distribution.

また、式（５）のｐ（Ｘ_t｜Λ_t）は、単一正規分布を用いて次の式（７）のようにモデル化される。

Also, p in formula _{(5) (X t | Λ} t) is modeled as the following equation (7) using a single normal distribution.

Therefore, the problem of sequentially estimating the probability distribution of the matrix Λ _t representing the disturbance based on the state space model 160 is that a sequence Λ _{0: t} that maximizes the posterior probability when the observed signal vector X _t is given. Reduce to the problem to estimate. The disturbance probability distribution estimation unit 200 performs this estimation based on the observation signal vector X _t and the state space model 160.

The disturbance probability distribution estimation unit 200 uses a technique called a particle filter when sequentially estimating the probability distribution of the matrix Λ _t representing the disturbance. This estimation method generates many localized state spaces (particles) in the state space, estimates the parameter probability distribution in each particle, and estimates the parameter probability distribution in the state space for each particle. It is a technique to express approximately using the probability distribution. In this method, initial parameters in a large number of particles are determined by random sampling or sampling from an initial distribution of the parameters. Then, the following processing is performed for each frame. That is, when a parameter is determined for each particle corresponding to a certain frame, the parameter of each particle is updated to that corresponding to the frame subsequent to that frame, and for each particle according to the likelihood of the update. Give weight. Then, the parameter of each particle corresponding to the subsequent frame is resampled according to the parameter probability distribution in the updated particle. Based on the resampled parameters, the parameters of each particle corresponding to the subsequent frame are determined. By performing the above processing for each frame, parameters for each particle are sequentially determined. The parameters in the state space are approximately expressed by the weighted sum of the parameters in the particles. That is, when the number of particles is J, a matrix composed of parameters corresponding to a matrix Λ _t representing disturbance in the j-th particle is a matrix Λ _t ^(j), and a weight for the particle is w _t ^(j) , The posterior probability distribution p (Λ _{0: t} | X _{0: t} ) of the sequence Λ _{0: t} shown in (5) is approximately expressed by the following equation (8).

Parameter generating unit 202, by HMM composition specifically called VTS (Vector Taylor Series) method, using the disturbance probability distributions estimated by the particle filter, the average of the feature vectors X _t of the observation signals in a plurality of particles Each has a function of calculating a vector and a covariance matrix (208).

  The clean speech estimation unit 204 estimates a clean speech parameter for each of a plurality of particles for each frame by a minimum mean square error (MMSE) estimation method, and calculates a weighted sum of these estimated parameters. It has a function of calculating the feature quantity 126 of the estimated clean speech. The clean speech estimation unit 204 further has a function of issuing a request 210 regarding the transition to the next frame to the disturbance probability distribution estimation unit 200.

  FIG. 6 is a block diagram showing the configuration of the disturbance probability distribution estimation unit 200. Referring to FIG. 6, disturbance probability distribution estimation section 200 receives a feature quantity 124 of the observation signal and a request 210 from clean speech estimation section 204, selects a frame to be processed, and observes the observation signal in that frame. Frame selection unit 220 for outputting the feature amount 124 of the image to the output destination corresponding to the frame, and estimating the disturbance probability distribution in the initial state by receiving the feature amount 124 of the observation signal for the first predetermined frame from the frame selection unit 220 Then, the disturbance initial distribution estimation unit 222 for determining the initial parameter of the disturbance in each particle and the feature amount 124 of the observation signal in the t (t ≧ 1) -th frame from the frame selection unit 220 are sequentially received. Includes a sequential calculation unit 224 for calculating a noise parameter for the particle and a weight for the particle. .

The disturbance initial distribution estimation unit 222 estimates a probability distribution (hereinafter, “disturbance initial distribution”) of a matrix Λ ₀ = (N ₀ , H ₀ , A ₀ ) representing the disturbance in the frame at time t = 0. At this time, the initial distribution of additive noise is estimated as follows.

The disturbance initial distribution estimation unit 222 first considers that the initial distribution of additive noise, that is, the probability distribution p (N ₀ ) of the initial value vector N ₀ of additive noise is a single normal distribution, and Estimate the initial distribution. When the average vector in the initial distribution of additive noise is μ _N and the covariance matrix is a matrix Σ _N , the initial distribution p (N ₀ ) of additive noise is expressed as the following equation (9).

外乱初期分布推定部２２２は、最初の所定フレーム分の区間の観測信号の特徴量ベクトルＸ_tが加法性雑音の成分のみからなるものとみなし、加法性雑音の初期分布の平均ベクトルμ_N、及び共分散行列Σ_Nを推定する。例えば０≦ｔ≦９の１０フレーム分の区間がこの区間に該当する場合、外乱初期分布推定部２２２は、平均ベクトルμ_N及び共分散行列Σ_Nをそれぞれ、次の式（１０）及び式（１１）によって算出する。ただし、ベクトルの右肩に付した「Ｔ」は転置を表す。

Disturbance initial distribution estimation unit 222, considers the feature vector X _t of the first predetermined number of frames interval of the observed signal consists only component of the additive noise, the mean vector mu _N of initial distribution of additive noise and, Estimate the covariance matrix Σ _N. For example, when a section of 10 frames of 0 ≦ t ≦ 9 corresponds to this section, the disturbance initial distribution estimation unit 222 calculates the average vector μ _N and the covariance matrix Σ _N using the following formulas (10) and ( 11). However, “T” attached to the right shoulder of the vector represents transposition.

次に外乱初期分布推定部２２２は、初期状態での各パーティクルにおける外乱を表す行列Λ₀ ^(j)を構成するベクトルＮ₀ ^(j)、行列Ｈ₀ ^(j)、及び行列Ａ₀ ^(j)を、式（１２）のように設定する。

Next, the disturbance initial distribution estimation unit 222 includes a vector N ₀ ^(j) , a matrix H ₀ ^(j) , and a matrix A ₀ ^(j) that constitute a matrix Λ ₀ ^(j) representing the disturbance in each particle in the initial state. Is set as shown in Expression (12).

すなわち、各パーティクルにおける加法性雑音のベクトルＮ₀ ^(j)を、初期分布ｐ（Ｎ₀）からのサンプリングによって生成し、各パーティクルにおける乗法性歪みの行列Ｈ₀ ^(j)及び残響の行列Ａ₀ ^(j)の各要素の値を０に設定する。

That is, a vector N ₀ ^(j) of additive noise in each particle is generated by sampling from the initial distribution p (N ₀ ), and a multiplicative distortion matrix H ₀ ^(j) and a reverberation matrix A ₀ in each particle. the value of each element of the ^(j) is set to 0.

Further, the disturbance initial distribution estimation unit 222 has a covariance matrix Σ of a vector N ₀ ^(j) , a matrix H ₀ ^(j) , and a matrix A ₀ ^(j) that constitute a matrix Λ ₀ ^(j) representing the disturbance in each particle. _N0 ^(j) , Σ _H0 ^(j) , and Σ _A0 ^(j) are set as in equation (13).

すなわち、各パーティクルにおける加法性雑音のベクトルＮ₀ ^(j)の共分散行列を、初期分布ｐ（Ｎ₀）の共分散行列に設定し、各パーティクルにおける乗法性歪みの行列Ｈ₀ ^(j)及び残響の行列Ａ₀ ^(j)の共分散行列の各要素を０に設定する。外乱初期分布推定部２２２は、式（１２）と式（１３）とに示す設定を、パーティクルｊ（１≦ｊ≦Ｊ）ごとに行なう。

That is, the covariance matrix of the additive noise vector N ₀ ^(j) in each particle is set to the covariance matrix of the initial distribution p (N ₀ ), and the multiplicative distortion matrix H ₀ ^(j) in each particle and Each element of the covariance matrix of the reverberation matrix A ₀ ^(j) is set to zero. The disturbance initial distribution estimation unit 222 performs the setting shown in Expression (12) and Expression (13) for each particle j (1 ≦ j ≦ J).

  The sequential calculation unit 224 receives a GMM sampling unit 226 for sampling the output parameter 140 of the GMM 130, an update unit 230 for updating the disturbance parameter in each particle in response to the observed signal feature 124 in the t-th frame, , A weight calculation unit 232 for calculating the weights for the updated particles, a re-sampling unit 234 for re-sampling the disturbance parameters in the particles based on the weights calculated by the weight calculation unit 232, and resampling The estimated disturbance distribution generation for determining the disturbance parameter for each particle and generating the estimated disturbance distribution 206 based on the disturbance parameter for the generated particle and the disturbance parameter for each particle in the (t-1) th frame. And a 236.

The updating unit 230 uses the extended Kalman filter configured based on the state space model 160 (FIG. 4) to update the noise parameter in the particle corresponding to the t−1 frame to the one corresponding to the t frame. Has function. The extended Kalman filter is a Kalman filter corresponding to a state space model including a nonlinear term as shown in Expression (1). The distribution update formulas of the extended Kalman filter in the present embodiment are shown in the following formulas (14) to (19). In these equations, “ _{t | t−1} ” is attached as a subscript to the parameter in the t-th frame predicted from the parameter corresponding to the t−1 frame.

ただし、式（１７）及び式（１８）のベクトルＳ^(j) _kt ^(j) _,tは、ｊ番目のパーティクルにおいてＧＭＭ１３０（図２参照）の出力パラメータベクトルＳ_kt,tに相当するパラメータである。また前述した通り、行列Σ_Wは、第ｔ−１フレームから第ｔフレームへの状態変化の際に外乱を表す行列Λ_tに生じる誤差の行列Ｗ_tの共分散行列を表す。

However, the vector S ^(j) _kt ^(j) _{, t} in the equations (17) and (18) is a parameter corresponding to the output parameter vector S _{kt, t} of the GMM 130 (see FIG. 2) in the j-th particle. . Further, as described above, the matrix Σ _W represents the covariance matrix of the error matrix W _t generated in the matrix Λ _t representing the disturbance at the time of the state change from the t−1 frame to the t frame.

The GMM sampling unit 226 samples a single normal distribution k _t ^(j) that is an element distribution from the mixture distribution in the GMM 130 (see FIG. 2) based on the mixture weight. The GMM sampling unit 226 further samples the output parameter vector S ^(j) _kt ^(j) _{, t} from the sampled element distribution k _t ^(j) according to the probability distribution, and supplies the sample to the update unit 230. Assuming that the mixing weight of the element distribution k _{t in} the GMM 130 is P _{S, st} ^(j) _{, kt} , the element distribution k _t ^(j) is a probability distribution with the mixing weight P _{S, st} ^(j) _{, kt} as the output probability. Follow. That is, it is obtained from the GMM 130 by sampling shown in the following equation (20).

要素分布ｋ_t ^(j)の平均ベクトルをベクトルμ_kt ^(j)とし、要素分布ｋ_t ^(j)の共分散行列を行列Σ_S,kt ^(j)とすると、ｊ番目のパーティクルにおけるＧＭＭ１３０の出力パラメータベクトルＳ^(j) _kt ^(j) _,tは、要素分布ｋ_t ^(j)から、次の式（２１）に示すサンプリングによって得られる。

If the average vector of the element distribution k _t ^(j) is the vector μ _kt ^(j) and the covariance matrix of the element distribution k _t ^(j) is the matrix Σ _{S, kt} ^(j) , the output of the GMM 130 at the j-th particle The parameter vector S ^(j) _kt ^(j) _{, t} is obtained from the element distribution k _t ^(j) by sampling shown in the following equation (21).

なお、フレーム選択部２２０はさらに、ＧＭＭサンプリング部２２６に対し、第ｔフレームにおけるＧＭＭの出力パラメータのサンプリングを要求する機能を持つ。

The frame selection unit 220 further has a function of requesting the GMM sampling unit 226 to sample the output parameters of the GMM in the t-th frame.

Weight calculation unit 232, a feature vector X _t of the observation signals in the t-th frame, the output parameter vector S of GMM130 in each particle of the t frame ^{_{(j) kt (j),}} t, and disturbance of parameter matrix Λ _{Based on t} ^(j) and the weight w _t-1 ^(j) for the particles in the ⁽ _t−1 ⁾ th frame, the particles in the tth frame are calculated using the calculation methods shown in the following equations (22) and (23). Has a function of calculating a weight w _t ^(j) .

なお、重みｗ_t ^(j)（１≦ｊ≦Ｊ）は、Σ_j=1〜Ｊｗ_t ^(j)＝１となるように正規化される。

The weights w _t ^(j) (1 ≦ j ≦ J) are normalized so that Σ _{j =} 1 to J w _t ^(j) = 1.

The re-sampling unit 234 has a function of re-sampling the disturbance parameter matrix Λ _t ^(j) in each particle corresponding to the time t according to the probability distribution of the disturbance parameter in the particle whose parameter has been updated. At this time, the re-sampling unit 234 does not re-sample the parameters from the probability distribution of the particles to which only a minute weight w _t ^(j) is given. On the other hand, from the probability distribution of particles with a large weight w _t ^(j) , the parameters are resampled a number of times according to the size of the weight w _t ^(j) , and the obtained parameters are Assign to the same number of particles as the number of resampling. However, the total number of resampling and the total number of particles are constant (J). This is because the weight assigned to each particle corresponds to the likelihood of the observed feature vector X _t as can be seen from Equation (22).

The estimated disturbance distribution generation unit 236 has a function of regenerating particles corresponding to the t-th frame by the Metropolis-Hastings algorithm. FIG. 7 is a block diagram illustrating the configuration of the estimated disturbance distribution generation unit 236. Referring to FIG. 7, the estimated disturbance distribution generation unit 236 represents the disturbance probability distribution in the state space model 160 using the disturbance probability distribution of each particle obtained by resampling by the resampling unit 234, and Based on the obtained probability distribution, the disturbance parameter in the particle corresponding to the (t-1) th frame is updated again to that corresponding to the tth frame using the extended Kalman filter shown in the above equations (14) to (19). The re-updating unit 262 and the weight for the re-updated particle (hereinafter referred to as “w _t ^{* (j)} ”) using the calculation method shown in the above formulas (22) and (23). a weight recalculation unit 264 for calculating weights w _t ^* or ^(j) for the weights w _t ^(j) and re-updated particle for resampled particles , An allowable probability calculation unit 266 for calculating an allowable probability ν used for determining whether or not the re-updated parameter is allowed, and a random number u within a closed interval from 0 to 1 by a predetermined random number generation method Based on the random number generation unit 268, the allowable probability ν, and the random number u, as a parameter in the particle corresponding to the t-th frame, one of the parameter in the resampled particle and the parameter in the reupdated particle is set. A parameter selection unit 270 for selection.

Acceptable probability calculation unit 266, a weight according to w _t ^(j) and the weights w _t ^{* (j)} from the following equation (24) has a function of calculating the permission probability [nu.

  The parameter selection unit 270 has a function of changing the disturbance parameter of the particle to a new parameter obtained by re-update if u is equal to or less than the allowable probability ν.

[Program structure]
As will be apparent from the following description, the preprocessing unit 104, the preprocessing acoustic model unit 106, and the search unit 110 of the speech recognition system 100 shown in FIG. 1 are all executed on computer hardware and the computer hardware. It can be realized by a program. FIG. 8 is a flowchart showing a control structure of a computer program for realizing disturbance component suppression processing performed by the disturbance component suppression unit 114 included in the preprocessing unit 104 according to the present embodiment.

Referring to FIG. 8, when the disturbance component suppression process is started, in step 302, an initial distribution corresponding to the value of each element of disturbance Λ ₀ in the initial state is estimated. That is, the calculation method shown in the above formula (10) and (11), calculates a parameter mean vector mu _N and covariance matrix sigma _N of initial distribution of additive noise represented by the formula (9) p (N ₀₎ To do. Further, according to the equations (12) and (13), the parameter vector N ₀ ^(j) (j = 1,..., J) is sampled from the initial distribution p (N ₀ ), and the initial additive noise of each particle is detected. To the typical parameters. At this time, the initial parameter matrix H ₀ ^(j) of multiplicative noise in each particle and the initial parameter matrix A ₀ ^(j) of reverberation are also set according to the equations (12) and (13), respectively. To do.

In step 304, the frame subject to disturbance suppression is shifted to the next frame. In step 306, the probability distribution parameter corresponding to the matrix representing the disturbance in the processing target frame is estimated using the particle filter. That is, the disturbance parameter matrix Λ _t ^{(j) of} each particle and the covariance matrix of the matrix Λ _t ^(j) are estimated, and the weight w ^(j) for each particle is determined. The processing in this step will be described later with reference to FIG.

In step 308, a disturbance parameter matrix defined for each particle lambda _t ^(j) in step 306, using its covariance matrix, the probability distribution of the feature vectors X _t of the observation signals of each particle (124) presume. Further, for each element distribution k (1 ≦ k ≦ K) constituting the GMM 130, an average vector μ _Xkt ^(j) _{, t} of a probability model of an observation signal in a particle and a covariance matrix Σ _{Xk, t} ^(j) calculate.

In step 310, the feature amount of clean speech in the t-th frame is estimated by the MMSE estimation method. That is, first, the MMSE estimation value vector {circumflex over (S) _} is calculated by the MMSE estimation method using the parameters obtained in the processing of step 306 and step 308 and is output as the estimated clean speech feature 126 (see FIG. 1). .

この式において、Ｐ（ｋ｜Ｘ_t，（ｊ））は、ｊ番目のパーティクルにおける、ＧＭＭ１３０内の要素分布ｋに対する混合重みを表す。混合重みＰ（ｋ｜Ｘ_t，（ｊ））は、次の数式により算出される。

In this equation, P (k | X _t , (j)) represents the mixing weight for the element distribution k in the GMM 130 in the j-th particle. The mixing weight P (k | X _t , (j)) is calculated by the following equation.

  In step 312, end determination is performed. That is, if the t-th frame is the final frame, the disturbance component suppression processing is terminated. Otherwise return to step 304.

  FIG. 9 is a flowchart showing a control structure of a program that realizes the disturbance probability distribution estimation process performed in step 306 (see FIG. 8). Referring to FIG. 9, when the disturbance probability distribution estimation process is started, in step 322, the disturbance probability of the particles in the t−1th frame using the extended Kalman filter expressed by equations (14) to (19). From the distribution, the disturbance probability distribution in the t-th particle is estimated.

In step 324, the weight w _t ^(j) for each particle in the t-th frame is calculated and normalized by the equations (22) and (23). In step 326, the number of re-sampling from each particle is determined based on the weight w _t ^(j) given to each particle, and the parameter is re-sampled based on the disturbance probability distribution of the particle. In step 328, the particles of the t-th frame are regenerated using the Metropolis-Hastings algorithm.

FIG. 10 is a flowchart showing details of the processing in step 328 (see FIG. 9). Referring to FIG. 10, when the processing in step 328 is started, in step 342, the disturbance probability distribution is re-updated using the parameters in the particles obtained by the resampling in step 326 (see FIG. 9). Do. That is, a new particle of the frame at time t is prepared, and the particle corresponding to the t-th frame is handled from the parameter corresponding to the particle of the (t-1) -th frame by the same process as the process in step 322 (see FIG. 9). Re-update to the parameter to be set, and set the parameter of the prepared particle. In step 344, the weight w _t ^{* (j)} for the particles prepared in step 342 is calculated and normalized by the same process as the process in step 324 shown in FIG.

In step 346, the allowable probability ν of the particles prepared in step 342 is determined by comparing the weight w _t ^(j) calculated in step 324 with the weight w _t ^{* (j)} calculated in step 344. Determine. In step 348, a random number u is generated by selecting an arbitrary value from the uniform set U _[0,1] consisting of values in the interval [0,1]. In step 350, the value of the random number u generated in step 348 is compared with the value of the allowable probability ν determined in step 346. If u is less than or equal to the allowable probability, the process proceeds to step 352. Otherwise, go to step 354. In step 352, the particles prepared in step 342 are allowed. That is, the parameter obtained by the resampling in step 326 is replaced with the parameter of the prepared particle, and the process is terminated. In step 354, the particles prepared in step 342 are rejected. That is, the prepared particles and their parameters are discarded, and the process ends.

[Operation]
The speech recognition system 100 according to the present embodiment operates as follows. First, the disturbance probability distribution estimation operation in the initial state by the disturbance probability distribution estimation unit 200 shown in FIG. 6 will be described. The measurement unit 112 shown in FIG. 1 receives the observation sound 122 from the sound source 102 and extracts the feature value X _t (124) of the observation signal. The extracted feature amount X _t (124) is given to the disturbance probability distribution estimation unit 200 shown in FIG. With reference to FIG. 6, the frame selection unit 220 of the disturbance probability distribution estimation unit 200 gives the first 10 frames of the feature amount X _t (124) to the disturbance initial distribution estimation unit 222. Disturbance initial distribution estimating unit 222 estimates the additive noise initial distribution p (N ₀₎ by the process shown in the above equation (9) to (11). Further, sampling shown in the above equations (12) and (13) is performed J times from the initial noise distribution p (N ₀ ). By this sampling, an initial parameter vector N ₀ ^(j) and a covariance matrix Σ _N0 ^(j) of noise in each particle are determined. The initial parameter matrix H ₀ ^{(j) of} multiplicative distortion and its covariance matrix Σ _H0 ^(j) are both set to 0, and the initial parameter matrix A ₀ ^{(j) of} reverberation and its covariance matrix Σ _A0 ^(j) are Both are set to 0. The disturbance probability distribution estimation unit 200 outputs these parameters as parameters of the estimated disturbance distribution 206 in the frame at time t = 0.

Next, the estimation operation of the estimated disturbance distribution 206 in the t-th frame (t ≧ 1) by the disturbance probability distribution estimation unit 200 will be described. Referring to FIG. 6, in response to processing start request 210 for the next frame, frame selection unit 220 provides observed signal feature quantity X _t (124) to update unit 230 and also to GMM sampling unit 226. , Request sampling of output parameters of the GMM in the t-th frame. Updating unit 230, in response thereto, to obtain the parameters of the estimated disturbance distribution 206 in each particle of the t-1 frame.

The GMM sampling unit 226 samples the output parameter vector S ^(j) _kt ^(j) _{, t} from the GMM 130. FIG. 11 schematically shows an outline of sampling of the output parameter vector S ^(j) _kt ^(j) _{, t} . For example, at the j-th particle, the element distribution k _t ^(j) (402) is sampled from the mixed normal distribution 400 in the GMM 130 with the probability according to the mixing weight. The GMM sampling unit 226 further samples the output parameter vector S ^(j) _kt ^(j) _{, t} (404) according to the output probability distribution represented by the element distribution k _t ^(j) (402). GMM sampling unit 226, an output parameter vector S in each particle of the total number _{^{J (j) kt (j)}} , t respectively, sampled at the above procedure, gives the updating section 230 shown in FIG.

FIG. 12 schematically shows an outline of parameter updating and resampling performed by the sequential calculation unit 224. In FIG. 12, certain disturbance parameters are distributed in the left-right direction, and the time advances from top to bottom. In FIG. 12, the particles are schematically shown by white circles and black circles. For example, a particle indicated by a white circle is a minute particle having a value of weight w _t ^(j) , and a particle indicated by a black circle is a particle having a large value of weight w _t ^(j) To do.

Referring to FIG. 12, it is assumed that state space 420 is approximately represented by particles corresponding to the (t-1) th frame. The updating unit 230 uses the extended Kalman filter expressed by the equations (14) to (19) to correspond the parameter matrix ^ Λ _t-1 ^(j) of the disturbance distribution in each particle in the state space 420 to the t-th frame. Update the parameter matrix ^ Λ _t ^(j) of the estimated disturbance distribution. As a result, each particle in the state space 420 is updated, and the state space 430 corresponding to the t-th frame is represented by the particle whose parameter has been updated.

Subsequently, the weight calculation unit 232 calculates the weight w _t ^(j) for each particle in the state space 430 using Expression (22) and Expression (23). The re-sampling unit 234 re-samples the disturbance parameter in the particle based on the weight w _t ^(j) . At this time, the re-sampling unit 234 first sets the number of re-sampling from each particle in the state space 430 for each particle according to w _t ^(j) . The number of samplings from a minute particle with a weight represented by a white circle is set to zero. In addition, the number of times of sampling from particles with a large weight represented by black circles is set to 1 to 3 according to the magnitude of the weight. Subsequently, based on the probability distribution of the disturbance in the particles in the state space 430, the disturbance parameter is resampled by the set number of times. In this way, particles representing a new state space 440 corresponding to the t-th frame are formed.

When such re-sampling by the re-sampling unit 234 is repeatedly performed, the disturbance parameters in many particles corresponding to a certain frame are obtained from the probability distribution of the disturbance parameters in a small number of particles corresponding to the previous frame. May be sampled. Therefore, the estimated disturbance distribution generation unit 236 prevents such a situation by newly generating parameters in the particles corresponding to the t-th frame using the Metropolis-Hastings algorithm. The re-update unit 262 illustrated in FIG. 7 re-updates the disturbance parameters of the particles in the state space 420 corresponding to the (t-1) th frame in accordance with the estimated disturbance distribution in the state space 440. The weight recalculation unit 264 calculates a weight w _t ^{* (j)} for the re-updated particle. Acceptable probability calculation unit 266, the weight w _t ^* for particles that are re-updated ^(j), based on the weight w _t ^(j) with respect to the resampled particles, calculates the permission probability [nu. The parameter selection unit 270 compares the allowable probability ν with the random number u in the interval [0, 1] generated by the random number generation unit 268. If the random number u is equal to or less than the allowable probability ν, the parameter selection unit 270 Replace the parameter with the parameter in the re-updated particle. Otherwise, discard the parameters in the re-updated particle.

By repeating the above operation for each frame, the parameter vector N _t ^(j) H _t ^(j) and the matrix At ^(j) of the estimated disturbance distribution 206 in each particle and the matrix A _t ^(j) covariance matrix ^{_{Σ Nt (j) Σ Ht (}} j) and sigma _At ^(j) is estimated. Disturbance probability distribution estimation unit 200, the parameter vector N _t of the estimated disturbance distribution 206 in each particle ^(j) H _t ^(j) and the matrix A _t ^(j), and the covariance matrix ^{_{Σ Nt (j) Σ Ht (}} j) And Σ _At ^(j) , the weight w _t ^(j) for each particle, and the feature vector X _t of the observation signal are given to the parameter generation unit 202 shown in FIG. 5 for each frame.

Referring to FIG. 5, the parameter generation unit 202 generates an average vector and a covariance matrix (208) of the probability model of the observation signal in each particle corresponding to the t-th frame by the VTS method. As a result, the probability distribution of the disturbance and the probability distribution of the observation signal are estimated for each particle. The clean speech estimation unit 204 calculates a clean speech MMSE estimated value vector ^{ circumflex over ⁽ S ⁾ _} ^(j) for each particle corresponding to the t-th frame by the MMSE estimation method. Further, using the MMSE estimated value vector ^{ circumflex over ⁽ S ⁾ } _t ^(j) and the weight w _t ^(j) , an estimated feature amount vector ^{ circumflex over ⁽ S ⁾ } at time _t is calculated and output to the search unit 110 shown in FIG. To do.

Searching unit 110 shown in FIG. 1, based on using the estimated feature vector ^ S _t of clean speech, the acoustic model held in the acoustic model for recognition section 109, and a language model stored in the language model 108 Then, a suitable target language word or the like is searched, and the result is output as a recognition output 128.

[Experiment]
In order to confirm the effect of the speech recognition system 100 according to the present embodiment, an experiment for estimating noise from the observed signal and an experiment for recognizing the observed signal were performed. Hereinafter, experimental methods and results will be described.

  In this experiment, the observed signal was generated by convolution of the impulse response of reverberation into the data of clean speech 1001 sentence recorded in the common database for speech recognition evaluation under noisy Japanese and artificially adding additive noise. . For the reverberation impulse response, impulse responses with a reverberation time of 0.3 seconds and 1.3 seconds recorded in the real environment speech and sound database were used. As noise to be added, factory noise and road construction noise recorded in the actual environment were used. In this experiment, a sample in which noise was not added and a sample in which noise was added to each of clean speech at an SNR (Signal-to-Noise Ratio) of 20 dB, 15 dB, and 10 dB were prepared. Each prepared sample was subjected to 23th-order log mel filter bank processing, and a vector having each component of the obtained 23th-order log mel spectrum as an element was generated and used as a feature quantity vector to be recognized.

  In the recognition experiment, for comparison, feature quantities used for the search were generated from the above samples by the following five processing methods including the disturbance component suppression processing method according to the present embodiment. That is, the feature quantity (Baseline) of the observed signal that is not subjected to the disturbance suppression process, the characteristic quantity subjected to the noise suppression process by ETSI Advanced front-end (ES 202) recommended by ETSI (European Telecommunications Standards Institute) (ETSI), an estimated feature amount (MMSE) obtained by conventional MMSE estimation, an estimated feature amount (EM) obtained by processing by the method described in Non-Patent Document 5, and a particle filter. This is an estimated feature amount obtained by the disturbance component suppression processing (Proposed).

A model having a mixture distribution number of 512 was used for the GMM 130 (see FIG. 2 ) when performing disturbance component suppression processing using a particle filter. In this processing, the covariance matrix of the error vector W _t, was set to _{Σ WN = Σ WH = Σ WA} = diag (0.01). In addition, the total number J of particles used in the processing was set to 20.

  A 39th-order MFCC (Mel Frequency Cepstrum Coefficient) (12th-order MFCC + C0 + Δ + ΔΔ) is used as a feature amount when performing speech recognition using the estimated clean speech after suppression. Further, a 16-state 20-mixed HMM was used for the recognition acoustic model 109 shown in FIG.

  When a commercially available CPU (Central Processing Unit) with a clock frequency of 3.2 GHz and a 32-bit clock was used for the processing in this recognition experiment, the time required for the processing was 0.8 times the real time of the observation signal. . That is, it became clear that the recognition process can be processed in real time.

  Tables 1 to 4 show the recognition accuracy obtained in the recognition experiment for each sample according to the above processing methods.

表１〜表４を参照して、パーティクルフィルタによる抑圧処理（Ｐｒｏｐｏｓｅｄ）を行なうことで、良好な単語認識精度が得られることが分かる。特に、残響１．３秒の観測信号においては、パーティクルフィルタによる抑圧処理（Ｐｒｏｐｏｓｅｄ）により、高い単語認識精度が得られることが分かる。

Referring to Tables 1 to 4, it can be seen that good word recognition accuracy can be obtained by performing suppression processing (Proposed) using a particle filter. In particular, it can be seen that a high word recognition accuracy can be obtained in the observation signal of reverberation 1.3 seconds by the suppression process (Proposed) using the particle filter.

  Based on the above experimental results, the disturbance component suppression processing according to the present embodiment improves speech recognition performance in an environment subject to distortion due to unsteady additive noise and reverberation, and enables real-time processing. Became clear.

[Modifications, etc.]
In the present embodiment, the processing by the particle filter is used for suppressing disturbance components. Therefore, acoustic model adaptation can be further performed before searching using the parameters of the estimated clean speech after noise suppression. With the acoustic model adaptation, an acoustic model suitable for the estimated clean speech can be used for the search. Therefore, it is expected that the recognition accuracy is improved.

  In this embodiment, GMM is used for the acoustic model for preprocessing. However, HMM may be used for the acoustic model for preprocessing. In this case, the state may be sampled according to the transition probability of the HMM prior to sampling the element distribution shown in the above equation (20).

  The embodiment disclosed herein is merely an example, and the present invention is not limited to the above-described embodiment. The scope of the present invention is indicated by each claim in the claims after taking into account the description of the detailed description of the invention, and all modifications within the meaning and scope equivalent to the wording described therein are intended. Including.

It is the schematic which shows the structure of the speech recognition system 100 which concerns on one embodiment of this invention. It is the schematic which shows the concept of GMM130. It is a figure which shows the disturbance factor 118 typically. It is the schematic which shows the concept of the state space model 160 of an observation signal. 3 is a block diagram illustrating a configuration of a disturbance component suppression unit 114. FIG. 3 is a block diagram showing a configuration of a disturbance probability distribution estimation unit 200. FIG. 4 is a block diagram illustrating a configuration of an estimated disturbance distribution generation unit 236. FIG. It is a flowchart which shows the control structure of a noise suppression process. It is a flowchart which shows the control structure of disturbance probability distribution estimation processing. It is a flowchart which shows the control structure of the sampling process by a Metropolis-Hastings algorithm. It is a figure which shows the outline | summary of the operation | movement which samples a parameter from GMM130. It is a figure which shows the outline | summary of the process by a particle filter.

Explanation of symbols

DESCRIPTION OF SYMBOLS 100 Speech recognition system 102 Sound source 104 Preprocessing part 106 Preprocessing acoustic model part 108 Language model part 109 Recognition acoustic model part 110 Search part 112 Measurement part 114 Disturbance distribution suppression part 116 Speaker 118 Disturbance factor 120 Clean voice 122 Observation sound 124 Features of observed signal 126 Features of estimated clean speech 130 GMM
132 learning data storage unit 134 model learning unit 136 GMM storage unit 160 state space model 200 disturbance probability distribution estimation unit 202 parameter generation unit 204 clean speech estimation unit 220 frame selection unit 222 disturbance initial distribution estimation unit 224 sequential calculation unit 226 GMM sampling unit 230 update unit 232 weight calculation unit 234 re-sampling unit 236 estimated disturbance distribution generation unit 262 re-update unit 264 weight re-calculation unit 266 allowable probability calculation unit 268 random number generation unit 270 parameter selection unit

Claims (4)

A disturbance component suppression device that suppresses a disturbance component of an observation signal obtained by observing target speech in an environment where additive noise and multiplicative distortion occur due to disturbance, wherein the disturbance component is a component due to additive noise, Includes components due to reverberation and multiplicative distortion,
Receiving feature amounts extracted from frames of a predetermined time length framed at predetermined intervals for the observation signal, the component due to the reverberation is regarded as additive disturbance, and using a particle filter having a plurality of particles A disturbance parameter estimation means for sequentially generating an estimation parameter of a probability distribution representing the disturbance for each frame;
Target speech estimation means for calculating an estimated feature amount of the target speech for each frame using a feature amount of the observed signal, the estimation parameter, and a predetermined acoustic model related to the target speech ;
Means for approximating a reverberation component included in the disturbance by a difference between the observed signal and a component due to the additive noise estimated by the disturbance parameter estimation means, and inputting the reverberation component to the disturbance parameter estimation means Disturbance component suppression device. The disturbance parameter estimation means includes:
An initial parameter setting means for setting an initial distribution of the disturbance and setting an initial parameter of a probability distribution representing the disturbance in each of the plurality of particles with a probability according to the initial distribution;
Based on the feature value of the acoustic model and the observed signal, regarded as a disturbance of additive components according to the reverberation, using an extended Kalman filter, wherein the estimated parameters of the first frame preceding in each particle respectively the Updating means for updating to one corresponding to a second frame following one frame;
The disturbance component suppressing device according to claim 1, further comprising weight calculating means for calculating the weight of each of the plurality of particles in the second frame. A computer program that, when executed by a computer, causes the computer to operate as the disturbance component suppressing device according to claim 1 or 2. The disturbance component suppressing device according to claim 1 or 2,
In response to the estimated feature amount of the target speech calculated by the disturbance component suppressing device, speech recognition related to the target speech is performed using a predetermined acoustic model related to the target speech and a predetermined language model related to a recognition target language. A speech recognition system, comprising: speech recognition means for performing.

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