Methods and apparatus for blind channel estimation based upon speech correlation structure

[0002] The present invention relates to methods and apparatus for processing speech signals, and more particularly for methods and apparatus for removing channel distortion in speech systems such as speech and speaker recognition systems.

[0003] Cepstral mean normalization (CMN) is an effective technique for removing communication channel distortion in automatic speaker recognition systems. To work effectively, the speech processing windows in CMN systems must be very long to preserve phonetic information. Unfortunately, when dealing with non-stationary channels, it would be preferable to use smaller windows that cannot be dealt with as effectively in CMN systems. Furthermore, CMN techniques are based on an assumption that the speech mean does not carry phonetic information or is constant during a processing window. When short windows are utilized, however, the speech mean may carry significant phonetic information.

[0004] The problem of estimating a communication channel affecting a speech signal falls into a category known as blind system identification. When only one version of the speech signal is available (i.e., the “single microphone” case), the estimation problem has no general solution. Oversampling may be used to obtain the information necessary to estimate the channel, but if only one version of the signal is available and no oversampling is possible, it is not possible to solve each particular instance of the problem without making assumptions about the signal source. For example, it is not possible to perform channel estimation for telephone speech recognition, when the recognizer does not have access to the digitizer, without making assumptions about the signal source.

[0005] One configuration of the present invention therefore provides a method for blind channel estimation of a speech signal corrupted by a communication channel. The method includes converting a noisy speech signal into either a cepstral representation or a log-spectral representation; estimating a temporal correlation of the representation of the noisy speech signal; determining an average of the noisy speech signal; constructing and solving, subject to a minimization constraint, a system of linear equations utilizing a correlation structure of a clean speech training signal, the correlation of the representation of the noisy speech signal, and the average of the noisy speech signal; and selecting a sign of the solution of the system of linear equations to estimate an average clean speech signal over a processing window.

[0006] Another configuration of the present invention provides an apparatus for blind channel estimation of a speech signal corrupted by a communication channel. The apparatus is configured to convert a noisy speech signal into either a cepstral representation or a log-spectral representation; estimate a temporal correlation of the representation of the noisy speech signal; determine an average of the noisy speech signal; construct and solve, subject to a minimization constraint, a system of linear equations utilizing a correlation structure of a clean speech training signal, the correlation of the representation of the noisy speech signal, and the average of the noisy speech signal; and select a sign of the solution of the system of linear equations to estimate an average clean speech signal over a processing window.

[0007] Yet another configuration of the present invention provides a machine readable medium or media having recorded thereon instructions configured to instruct an apparatus including at least one of a programmable processor and a digital signal processor to: convert a noisy speech signal into a cepstral representation or a log-spectral representation; estimate a temporal correlation of the representation of the noisy speech signal; determine an average of the noisy speech signal; construct and solve, subject to a minimization constraint, a system of linear equations utilizing a correlation structure of a clean speech training signal, the correlation of the representation of the noisy speech signal, and the average of the noisy speech signal; and select a sign of the solution of the system of linear equations to estimate an average clean speech signal over a processing window.

[0008] Configurations of the present invention provide effective and efficient estimations of speech communication channels without removal of phonetic information.

[0009] Further areas of applicability of the present invention will become apparent from the detailed description provided hereinafter. It should be understood that the detailed description and specific examples, while indicating the preferred embodiment of the invention, are intended for purposes of illustration only and are not intended to limit the scope of the invention.

[0010] The present invention will become more fully understood from the detailed description and the accompanying drawings, wherein:

[0011] FIG. 1 is a functional block diagram of one configuration of a blind channel estimator of the present invention.

[0012] FIG. 2 is a block diagram of a two-pass implementation of a maximum likelihood module suitable for use in the configuration of FIG. 1.

[0013] FIG. 3 is a block diagram of a two-pass GMM implementation of a maximum likelihood module suitable for use in the configuration of FIG. 1.

[0014] FIG. 4 is a functional block diagram of a second configuration of a blind channel estimator of the present invention.

[0015] FIG. 5 is a flow chart illustrating one configuration of a blind channel estimation method of the present invention.

[0016] The following description of the preferred embodiment(s) is merely exemplary in nature and is in no way intended to limit the invention, its application, or uses.

[0017] As used herein, a “noisy speech signal” refers to a signal corrupted and/or filtered by a communication channel. Also as used herein, a “clean speech signal” refers to a speech signal not filtered by a communication channel, i.e., one that is communicated by a system having a flat frequency response, or a speech signal used to train acoustic models for a speech recognition system. An “average clean version of a noisy speech signal” refers to an estimate of the noisy speech signal with an estimate of the corruption and/or filtering of the communication channel removed from the speech signal.

[0018] In one configuration of a blind channel estimator 10 of the present invention and referring to FIG. 1, a speech communication channel 12 is estimated and compensated utilizing a stored speech correlation structure Â(τ) 14. Blind channel estimator 10 as shown in FIG. 1 is representative of a portion of a speech recognition system, where the output of channel 12 is a noisy speech signal g(t)=s(t)*h(t), where s(t) represents a “clean” speech signal obtained using the output of microphone or audio processor 16 or via a filter having a flat frequency response, and h(t) represents the channel 12 filter. The signal represented by g(t) is converted into a signal Y(t)=S(t)+H(t) in the cepstral (or log spectral) domain by cepstral analysis module 18 (or by a log spectral analysis module, not shown).

[0019] Let S(t) be a “clean” speech signal represented in the cepstral (or log spectral) domain. Under the assumption that the inter-frame time correlation of clean speech is a decreasing function of τ:

[0020] E[S(t)S^T(t+τ)]=𝑓_τ(E[S(t)S(t)S^T(t)]),  (1)

[0021] 𝑓_τ is approximated by a time-invariant linear filter:

[0022] 𝑓_τ(E[S(t)S(t)S^T(t)])=A(τ)E[S(t)S^T(t)].  (2)

[0023] An estimate Â(τ) of the matrix A(τ) is derived from a clean speech training signal s(t) by performing a cepstral analysis (i.e., obtaining S(t) in the cepstral domain) and then performing a correlation written as: $\begin{array}{cc}E\left[S\left(t\right)S^T\left(t+\tau \right)\right]\approx \frac{1}{N}{\int }_0^{N}S\left(t+\omega \right)S^T\left(t+\tau +\omega \right)d\omega ,& \left(3\right)\end{array}$

[0024] averaging the ratio of E[S(t)S^T(t+τ)] and E[S(t)S^T(t)] (i.e., a correlation at delay τ and at zero delay): $\begin{array}{cc}A\left(t,\tau \right)=\frac{E\left[S\left(t\right)S^T\left(t+\tau \right)\right]}{E\left[S\left(t\right)S^T\left(t\right)\right]},& \left(4\right)\end{array}$

[0025] and integrating over the training database: $\begin{array}{cc}\hat{A}\left(\tau \right)=E\left[A\left(\tau \right)\right]\approx \frac{1}{N}{\int }_0^{T}A\left(t,\tau \right)dt& \left(5\right)\end{array}$

[0026] where the integral in equation 3 is carried out over the N samples of the processing window, and the integral in equation 5 is carried out over the whole training database. The computational steps described by equations 3 to 5 are carried out on a clean speech training signal obtained in an essentially noise-free environment so that a signal essentially equivalent to s(t) is obtained. Estimate Â(τ) obtained from this signal is stored in correlation structure module 14 prior to commencement of operation of blind channel estimator 10 with noisy channel 12.

[0027] For channel estimation, it is desirable to use small time lags for which the assumption in equation 1 is well verified, i.e., has small relative error, but not so small a time lag such that the speech signal correlation does not dominate the communication channel correlation.

[0028] Noisy speech signal Y(t) produced by cepstral analysis module 18 (or a corresponding log spectral module) is observed in the cepstral domain (or the corresponding log-spectral domain). Noisy speech signal Y(t) is written:

[0029] Y(t)=S(t)+H(t),  (6)

[0030] where S(t) is the cepstral domain representation of the original, clean speech signal s(t) and H(t) is the cepstral domain representation of the time-varying response h(t) of communication channel 12. The correlation of the observed signal Y(t) is then determined by correlation estimator 20. Let us represent the correlation function of signal Y(t) with a time-lag τ version Y(t+τ) (or equivalently, Y(t−τ)) as C_Y(τ), where C_Y(τ)=E[Y(t)Y^T(t+τ)].

[0031] Linear system solver module 22 derives a term A from the correlation C_Yproduced by correlation estimator 20 and correlation structure Â(τ) stored in correlation structure module 14:

[0032] A=(I−Â(τ))⁻¹(C_Y(τ)−Â(τ)C_Y(0)).  (7)

[0033] Also, averager module 24 determines a value b based on the output Y(t) of cepstral analysis module 18:

[0034] b=E[Y(t)],  (8)

[0035] and linear equation solver 22 solves the following system of equations for μ_s:

[0036] μ_sμ_s^T=bb^T−A=B, and  (9)

[0037] μ_s+H=b.  (10)

[0038] Systems of equations 9 and 10 are overdetermined, meaning that the number of separate equations exceeds the number of unknowns. Thus, in blind channel estimator 10, the system of equations is solved as a minimization problem, such as a minimum mean square error problem. Equation 10 is solved for μ_s=ŝ, where μ_sis an estimate of the average value of the mean speech signal without the channel corruption or filtering over a processing window, with linear system solver 22 minimizing $\begin{array}{cc}\underset{{\mu }_s}{\min }{\Vert {\mu }_s{\mu }_s^T-B\Vert }^2.& \left(11\right)\end{array}$

[0039] (The estimate {circumflex over (μ)}_sin one configuration is not used for speech recognition, as the processing window for channel estimation is longer, e.g., 40-200 ms, than is the window used for speech recognition, e.g., 10-20 ms. However, in this configuration, {circumflex over (μ)}_sis used to estimate $\hat{H},\mathrm{where}\hat{H}=\frac{1}{T}\sum ^{}Y\left(t\right)-{\hat{\mu }}_s,$

[0040] where the summation is over the processing window (e.g., 200 ms), and then S(t) is used for recognition in a shorter processing window, where Ŝ(t)=Y(t)−Ĥ.) In this configuration, S(t) represents clean speech over a shorter processing window, and is referred to herein as “short window clean speech.”

[0041] In one configuration of the present invention, an efficient minimization is performed by linear system solver 22 by setting

[0042] μ_s=±λ₁p₁,  (12)

[0043] where λ₁is the largest eigenvalue of B and p₁is the corresponding eigenvector. The solution to equation 12 is obtained in this configuration by searching for the eigenvector corresponding to the largest eigenvalue (in absolute value). This is a sub case of diagonalization problem for non-symmetric real matrices. Methods are known for solving this type of problem, but their precision is bounded by the ratio between the largest and smallest eigenvalues, i.e., the numerical methods are more stable for larger eigenvalue differences. Experimentally, the largest and second largest eigenvalues in configurations of the present invention have been found to differ by between about one and two orders of magnitude. Therefore, adequate stability is provided, and it is safe to assume that there exists one eigenvector that minimizes the cost function much better than any others. This eigenvector provides an estimate of the average clean speech μ_sover the processing window.

[0044] Because the speech estimate is obtained in modulus, a heuristic is utilized to obtain the correct sign. In blind channel estimator 10, acoustic models are used by maximum likelihood estimator module 26 to determine the sign of the solution to equation 12. For example, the maximum likelihood estimation is performed in two decoding passes, or with speech and silence Gaussian mixture models (GMMs).

[0045] In one configuration of a two-pass maximum likelihood estimator block 26 and referring to FIG. 2, Y(t) is input to two estimator modules 52, 54. Estimator module 52 also receives {circumflex over (μ)}_sas input, and estimator module 54 also receives −{circumflex over (μ)}_sas input. The result from estimator module 52 is Ŝ⁺(t), while the result from estimator module 54 is Ŝ⁻(t). These results are input to full decoders 56 and 58, respectively, which perform speech recognition. The output of full decoders 56 and 58 are input to a maximum likelihood selector module 60, which selects, as a result, words output from full decoders 56 and 58 using likelihood information that accompanies the speech recognition output from decoders 56 and 58. In one configuration not shown in FIG. 2, maximum likelihood selector module 60 outputs Ŝ(t) as either Ŝ⁺(t) or −Ŝ⁻(t). The output of S(t) is either in addition to or as an alternative to to the decoded speech output of decoder modules 56 and 58, but is still dependent upon the likelihood information provided by modules 56 and 58.

[0046] As an alternative to two-pass maximum likelhood determination block 26 of FIG. 2, a configuration of a two-pass GMM maximum likelihood decoding module 26A is represented in FIG. 3. In this configuration, estimates {circumflex over (μ)}_sand −{circumflex over (μ)}_sare input to speech and silence GMM decoders 72 and 74 respectively, and a maximum likelihood selector module 76 selects from the output of GMM decoders 72 and 74 to determine Ŝ(t), which is output in one configuration. In one configuration and as shown in FIG. 3, the output of maximum likelihood selector module 76 is provided to full speech recognition decode module 78 to produce a resulting output of decoded speech.

[0047] In another configuration of a blind channel estimator 30 of the present invention and referring to FIG. 4, the same minimization is utilized in linear system solver module 22, but a minimum channel norm module 32 is used to determine the sign of the solution. In blind channel estimator 30, the sign of μ_s=Ŝ(t) that minimizes the norm of the channel cepstrum ∥H(t)∥²=∥Y−μ_s∥²is selected as the correct sign of the solution ±μ_s. This solution for the sign is based on the assumption that, on average, the norm of the channel cepstrum is smaller than the norm of the speech cepstrum, so that the sign of ±μ_sthat minimizes ∥H(t)∥²=∥Y−μ_s∥²is selected as the speech signal Ŝ(t).

[0048] The estimated speech signal Ŝ(t) in the cepstral domain (or log-spectral domain) is suitable for further analysis in speech processing applications, such as speech or speaker recognition. The estimated speech signal may be utilized directly in the cepstral (or log-spectral) domain, or converted into another representation (such as the time or frequency domain) as required by the application.

[0049] In one configuration of a blind channel estimation method 100 of the present invention and referring to FIG. 5, a method is provided for blind channel estimation based upon a speech correlation structure. A correlation structure Â(t) is obtained 102 from a clean speech training signal s(t). The computational steps described by equations 3 to 5 are carried out by a processor on a clean speech training signal obtained in an essentially noise-free environment so that the clean speech signal is essentially equivalent to s(t).

[0050] A noisy speech signal g(t) to be processed is then obtained and converted 104 to a cepstral (or log-spectral) domain representation Y(t). Y(t) is then used to estimate 106 a correlation C_Y(τ) and to determine 108 an average b of the observed signal Y(t). The system of linear equations 9 and 10 is constructed and solved 110 subject to the minimization constraint of equation 11. A maximum likelihood method or norm minimalization method is utilized to select or determine 112 the sign of the solution, which thereby produces an estimate of the average clean speech signal over the processing window.

[0051] Better results are obtained with configurations of the present invention when the speech source and the communication channel more closely meet four conditions:

[0052] 1. S(t) and H(t) are two independent stochastic processes.

[0053] 2. E[S(t+τ)]=E[S(t)], i.e., S(t) is a short-term stationary process.

[0054] 3. The channel H(t) is constant within the processing window, so that H(t)=H, i.e., short-term invariance applies.

[0055] 4. The correlation structure of the speech source satisfies the time-invariant linear filter model, i.e., E[S(t)S^T(t+τ)]=A(τ)E[S(t)S^T(t)].

[0056] These conditions are considered to be sufficiently satisfied for small time-lags (short term structure). However, the second condition is not strictly satisfied when using the usual expectation estimator: $\begin{array}{cc}E\left[S\left(t\right)S^T\left(t+\tau \right)\right]=\frac{1}{N-\tau }\sum _{i=1}^{N-\tau }S\left(i\right)S^T\left(i+\tau \right).& \left(13\right)\end{array}$

[0057] Therefore, one configuration of the present invention utilizes a circular processing window: $\begin{array}{cc}E\left[S\left(t\right)S^T\left(t+\tau \right)\right]=\frac{1}{N-\tau }\sum _{i=1}^{N-\tau }S\left(i\right)S^T\left(i+\tau \right)+\frac{1}{\tau }\sum _{i=1}^{\tau }S\left(N-i\right)S^T\left(i\right).& \left(14\right)\end{array}$

[0058] Also, in one configuration of the present invention, to more closely satisfy the correlation structure condition, a speech presence detector is utilized to ensure that silence frames are disregarded in determining correlation, and only speech frames are considered. In addition, short processing windows are utilized to more closely satisfy the short-term invariance condition. One configuration of the present invention thus provides a speech detector module 19 to distinguish between the presence and absence of a speech signal, and this information is utilized by correlation estimator module 20 and averager module 24 to ensure that only speech frames are considered.

[0059] In one configuration of the present invention, the methods described above are applied in the cepstral domain. In another configuration, the methods are applied in the log-spectral domain. In one configuration, to ensure the precision of a diagonalization method utilized to solve the mean square error problem, the dynamic range of coefficients in the cepstral or log-spectral domain are made comparable to one another. (There are, in general, a plurality of coefficients because the cepstral or log-spectral features are vectors.) For example, in one configuration, cepstral coefficients are normalized by subtracting out a long-term mean and the covariance matrix is whitened. In another configuration, log-spectral coefficients are used instead of cepstral coefficients.

[0060] Cepstral coefficients are utilized for channel removal in one configuration of the present invention. In another configuration, log-spectral channel removal is performed. Log-spectral channel removal may be preferred in some applications because it is local in frequency.

[0061] In one configuration of the present invention, a time lag of four frames (40 ms) is utilized to determine incoming signal correlation. This configuration has been found to be an effective compromise between low speech correlation and low intrinsic hypothesis error. More specifically, if the processing window is excessively long, H(t) may not be constant, whereas if the processing window is excessively short, it may not be possible to get good correlation estimates.

[0062] Configurations of the present invention can be realized physically utilizing one or more special purpose signal processing components (i.e., components specifically designed to carry out the processing detailed above), general purpose digital signal processor under control of a suitable program, general purpose processors or CPUs under control of a suitable program, or combinations thereof, with additional supporting hardware (e.g., memory) in some configurations. For real-time speech recognition (for example, speech control of vehicles or type-as-you-speak computer systems), a microphone or similar transducer and an audio analog-to-digital (ADC) converter would be used to input speech from a user. Instructions for controlling a general purpose programmable processor or CPU and/or a general purpose digital signal processor can be supplied in the form of ROM firmware, in the form of machine-readable instructions on a suitable medium or media, not necessarily removable or alterable (e.g., floppy diskettes, CD-ROMs, DVDs, flash memory, or hard disk), or in the form of a signal (e.g., a modulated electrical carrier signal) received from another computer. An example of the latter case would be instructions received via a network from a remote computer, which may itself store the instructions in a machine-readable form.

[0063] A further mathematical analysis of the configuration described herein follows.

[0064] A speech signal corrupted by a communication communication channel observed in a cepstral domain (or a log-spectral domain) is characterized by equation 6 above. The correlation at time t with time lag τ of a signal X is given by:

[0065] C_X(τ)=E[X(t)X^T(t+τ)].  (15)

[0066] Assuming the independence, short-term stationarity, and short-term invariance conditions defined in the text above, the correlation of the observed signal can be written:

[0067] C_Y(τ)=C_S(τ)+μ_sH^T+Hμ_S^T+HH^T,  (16)

[0068] where μ_s=E[S(t)]. Equations 7 and 8 above are derived by assuming the short-term linear correlation structure condition defined in the text above.

[0069] An efficient minimization is derived by considering the following minimization problem in the N₂norm: $\begin{array}{cc}{\underset{X}{\min }\Vert {\mathrm{XX}}^T-B\Vert }^2,& \left(17\right)\end{array}$

[0070] where X=[x₁x₂. . . x_n]^Tand B=(b_i,j)_{i,jε1, . . . ,n}. Provided that B is diagonalizable, we can write B=PΛP*, where Λ=diag{λ₁. . . λ_n} is a diagonal matrix and P={p₁, . . . , p_n} is a unitary matrix. Consider the eigenvalues λ₁. . . λ_nto be sorted in increasing order λ₁≧ . . . ≧λ_n. It can be shown that: $\begin{array}{cc}{{\underset{X}{\min }\Vert {\mathrm{XX}}^T-B\Vert }^2\sim \underset{Y}{\min }\Vert {\mathrm{YY}}^T-\Lambda \Vert }^2,& \left(18\right)\end{array}$

[0071] with Y=P^TX. It can also be written: $\begin{array}{cc}{\Vert {\mathrm{YY}}^T-\Lambda \Vert }^2=\sum _i^n{\left(y_i^2-{\lambda }_i\right)}^2+\sum _i^{}\sum _{j\ne i}^{}{\left(y_i,y_j\right)}^2.& \left(19\right)\end{array}$

[0072] By taking partial derivatives, we have: $\begin{array}{cc}\frac{\partial {\Vert {\mathrm{YY}}^T-\Lambda \Vert }^2}{\partial y_k}=4y_k\left(\sum _i^{}{y}_i^2-{\lambda }_k\right).& \left(20\right)\end{array}$

[0073] By setting the derivatives to zero, we obtain: $\begin{array}{cc}4y_k\left(\sum _i{y}_i^2-{\lambda }_k\right)=0,\forall k=1\dots n.& \left(21\right)\end{array}$

[0074] Since it has been assumed that λ₁> . . . >λ_n, from the previous equation, it follows that at most one coefficient among y₁. . . y_nis nonzero. By contradiction, assume that ∃i₁≠i₂:y_i≠0, y_i≠0, then we would obtain: $\begin{array}{cc}\sum _i{y}_i^2={\lambda }_{\mathrm{i1}},& \left(22\right)\\ \sum _i{y}_i^2={\lambda }_{\mathrm{i2}},& \left(23\right)\end{array}$

[0075] and λ_i≠λ_i, which is impossible. Moreover, given that Y is a non-zero vector, we have: $\begin{array}{cc}\{\begin{array}{c}{y}_{i_0}=\pm {\lambda }_{i_0}\\ y_i=0\forall i\ne i_{i_0}\end{array}& \left(24\right)\end{array}$

[0076] Therefore, we conclude that ∥YY^T−Λ∥²=Σ_i≠iλ_i²and the solution that minimizes ∥YY^T−Λ∥²is i₀=1. This also implies that the minimization problem has two solutions X=±λ₁p₁, where λ₁is the largest eigenvalue of B and p₁is the corresponding eigenvector.

[0077] Configurations of the present invention provide effective estimation of a communication channel corrupting a speech signal. Experiments utilizing the methods and apparatus described herein have been found to be more effective that standard cepstral mean normalization techniques because the underlying assumptions are better verified. These experiments also showed that static cepstral features, with channel compensation using minimum norm sign estimation, provide a significant improvement compared to CMN. For maximum likelihood sign estimation, it is recommended that one consider the channel sign as a hidden variable and optimize for it during the expectation maximum (EM) algorithm, while jointly estimating the acoustic models.

[0078] In general, for a configuration of the present invention utilizing the cepstral domain throughout, there is a corresponding configuration of the present invention that utilizes the cepstral domain throughout. Once a design choice of one or the other domain is made, it should be used consistently throughout the configuration to avoid the need for additional conversions from one domain to the other.

[0079] The description of the invention is merely exemplary in nature and, thus, variations that do not depart from the gist of the invention are intended to be within the scope of the invention. Such variations are not to be regarded as a departure from the spirit and scope of the invention.