Signal processing device, signal processing method, and signal processing program

Abstract

To achieve both fast convergence and low distortion output signals without manual control of step size, even when a gain of an acoustic impulse response approximated by an adaptive filter is less than 1.SOLUTION: A signal processor for this application includes: first input means for inputting a first mixed signal in which a first signal and a second signal are mixed; second input means for inputting a second mixed signal in which a third signal correlated with the first signal and a fourth signal correlated with the second signal are mixed; a first adaptive filter for filtering the second mixed signal to generate an estimate of the second signal; a first subtraction unit for generating an estimate of the first signal from the first mixed signal and the estimate of the second signal; and an estimation unit that estimates a ratio of an amplitude or power of the first signal and the second signal as a first mixing ratio, using the estimated value of the first signal, the estimated value of the second signal, the second mixed signal, the first mixed signal, and a coefficient of the first adaptive filter, to control the first adaptive filter by using the first mixing ratio.SELECTED DRAWING: Figure 1

Description

[Technical Field]

[0001] The present invention relates to a signal processing technique for eliminating noise, interference signals, echoes, and the like that are mixed in a signal. [Background technology]

[0002] Speech signals input from a microphone, handset, or the like are often superimposed with background noise, posing a significant problem in speech coding and speech recognition. Patent Documents 1 and 2 disclose a two-input noise canceller using two adaptive filters as a signal processing device designed to cancel acoustically superimposed noise. The device inputs a noisy signal (a signal containing a mixture of a desired signal and noise) and a reference signal (mainly including a signal correlated with the noise), cancels some or all of the noise, and outputs an enhanced signal (a signal in which the desired signal is enhanced). A step-size calculation unit calculates the coefficient update step size of the second adaptive filter using the signal-to-noise ratio of the noisy signal estimated using the first adaptive filter. The first adaptive filter operates in the same way as the second adaptive filter, but the coefficient update step size of the first adaptive filter is set to a larger value than the coefficient update step size of the second adaptive filter. Therefore, the output of the first adaptive filter has high adaptability to environmental changes but is inferior to the second adaptive filter in noise estimation accuracy.

[0003] The step size calculation unit evaluates the signal-to-noise ratio of the noisy signal estimated using the first adaptive filter, and when the speech signal is louder than the noise, it determines that interference from the speech signal is large and provides a small coefficient update step size to the second adaptive filter. Conversely, when the speech signal is quieter than the noise, it determines that interference from the speech signal is small and provides a large coefficient update step size to the second adaptive filter. In this way, by controlling the second adaptive filter with the coefficient update step size provided by the step size calculation unit, sufficient adaptability to environmental changes and low distortion in the signal after noise cancellation are output.

[0004] Patent Document 3 discloses a configuration in which the first adaptive filter is omitted from the configurations of Patent Documents 1 and 2. The signal-to-noise ratio is approximated by the ratio between the desired signal (such as speech) estimated using a second adaptive filter and the output of the second adaptive filter, and the second adaptive filter itself is controlled using a step size calculated based on the signal-to-noise ratio. Furthermore, Patent Document 3 discloses a noise canceller configuration that extends the configurations of Patent Documents 1 and 2, and also cancels speech signals mixed into noise when crosstalk due to speech signals is present, which is a significant influence of speech signals mixed into noise at the input of a two-noise input device. In addition to the configurations of Patent Documents 1 and 2, Patent Document 3 includes a third adaptive filter that cancels crosstalk from a reference signal. To accurately cancel noise from the speech signal input, a second step size calculation unit calculates a coefficient update step size and controls the third adaptive filter.

[0005] That is, the noise cancellers in Patent Documents 1 to 3 control updating of the adaptive filter coefficients using a signal-to-noise ratio estimated using the signal after noise cancellation and the adaptive filter output. By using a small step size when the signal-to-noise ratio is high and a large step size when the signal-to-noise ratio is low, they achieve both fast convergence and low-distortion output signals.

[0006] However, in the noise cancellers of Patent Documents 1 to 3, the adaptive filter coefficients are not updated at all. This is because the initial values ​​of the adaptive filter coefficients are usually set to zero. An adaptive filter with zero coefficients outputs zero. Because this is the denominator of the estimated signal-to-noise ratio, the estimated signal-to-noise ratio becomes an extremely large value, and the corresponding step size is set to zero. A step size of zero means that no coefficient update is performed. To avoid this, the step size must be forcibly set to a non-zero value immediately after the coefficient update begins. However, no clear design method is disclosed regarding what value the step size should actually be set to or how long it should remain non-zero. In other words, manual control of the step size is required to achieve both fast convergence and low-distortion output signals in a two-input noise canceller.

[0007] Patent Document 4 discloses the configuration of a two-input noise canceller that does not require manual control of the step size and achieves both fast convergence and low-distortion output signals. A new signal-to-noise ratio, defined as the ratio of the post-noise-cancellation signal to the reference signal, is used immediately after the device is activated to control the update of the adaptive filter coefficients, solving the problem of non-coefficient update. When the adaptive filter coefficients grow, they are switched to the signal-to-noise ratio disclosed in Patent Document 3. The coefficient growth is evaluated when the new signal-to-noise ratio becomes sufficiently close to the conventional signal-to-noise ratio. When crosstalk is present, a third adaptive filter is introduced to cancel the speech signal from the noise input signal, and the step size is controlled using the same principle as the second adaptive filter. [Prior art documents] [Patent documents]

[0008] [Patent Document 1] Japanese Patent Application Publication No. 10-215193 [Patent Document 2] Japanese Patent Application Laid-Open No. 2000-172299 [Patent Document 3] International Publication No. 2012 / 046582 [Patent Document 4] International Publication No. 2019 / 092798 Summary of the Invention [Problem to be solved by the invention]

[0009] However, in the signal processing device described in Patent Document 4, if the gain of the acoustic impulse response approximated by the adaptive filter is less than 1, the estimated signal-to-noise ratio is not switched. This is because the new signal-to-noise ratio is not sufficiently close to the previous signal-to-noise ratio with an acoustic impulse response gain less than 1. As a result, the accuracy of the signal-to-noise ratio estimation is low, and it is not possible to achieve both fast convergence and a low-distortion output signal.

[0010] An object of the present invention is to provide a technique for solving the above problems. [Means for solving the problem]

[0011] In order to achieve the above object, the device according to the present invention comprises: a first input means for inputting a first mixed signal in which the first signal and the second signal are mixed; a second input means for inputting a second mixed signal in which a third signal correlated with the first signal and a fourth signal correlated with the second signal are mixed; a first adaptive filter that filters the second mixture signal to generate an estimate of the second signal; a first subtraction unit that generates an estimate of the first signal from the first mixed signal and an estimate of the second signal; an estimation unit that estimates a ratio of amplitudes or powers of the first signal and the second signal as a first mixture ratio using the estimated value of the first signal, the estimated value of the second signal, the second mixture signal, and a coefficient of the first adaptive filter; Equipped with The first mixing ratio is used to control the first adaptive filter.

[0012] In order to achieve the above object, the method according to the present invention comprises: Input the first mixed signal, which is a mixture of the first and second signals, a second mixed signal in which a third signal correlated with the first signal and a fourth signal correlated with the second signal are mixed is input; filtering the second mixed signal to generate an estimate of the second signal; generating an estimate of the first signal from the first mixed signal and an estimate of the second signal; estimating an amplitude or power ratio between the first signal and the second signal as a first mixture ratio using the estimated value of the first signal, the estimated value of the second signal, the second mixture signal, and a coefficient of the first adaptive filter; The first mixing ratio is used to control generation of an estimate of the second signal.

[0013] In order to achieve the above object, the program according to the present invention comprises: On the computer, inputting a first mixed signal in which a first signal and a second signal are mixed; inputting a second mixed signal in which a third signal correlated with the first signal and a fourth signal correlated with the second signal are mixed; filtering the second mixture signal to generate an estimate of the second signal; generating an estimate of the first signal from the first mixed signal and an estimate of the second signal; a step of estimating a ratio of amplitudes or powers of the first signal and the second signal as a first mixture ratio using the estimated value of the first signal, the estimated value of the second signal, the second mixture signal, and a coefficient of the first adaptive filter; using the first mixing ratio to control generation of an estimate of the second signal; Execute the following. [Effects of the Invention]

[0014] According to the present invention, a signal processing device can be obtained that can achieve both fast convergence and low distortion output signals without manually controlling the step size, even if the gain of the acoustic impulse response that the adaptive filter approximates is less than 1. [Brief explanation of the drawings]

[0015] [Figure 1] 1 is a block diagram showing a configuration of a signal processing device according to a first embodiment of the present invention. [Figure 2] FIG. 10 is a block diagram showing the configuration of a signal processing device according to a second embodiment of the present invention. [Figure 3] FIG. 10 is a block diagram showing a first configuration of an estimation unit according to a second embodiment of the present invention. [Figure 4] FIG. 10 is a diagram showing time transitions of values ​​according to the second embodiment of the present invention. [Figure 5] FIG. 10 is a block diagram showing a second configuration of an estimation unit according to a second embodiment of the present invention. [Figure 6] FIG. 10 is a block diagram showing a configuration of a signal processing device according to a third embodiment of the present invention. [Figure 7]FIG. 10 is a block diagram showing a first configuration of an estimation unit according to a third embodiment of the present invention. [Figure 8] FIG. 11 is a block diagram showing a second configuration of an estimation unit according to the third embodiment of the present invention. [Figure 9] 1 is a block diagram showing the configuration of a computer according to a first embodiment of the present invention. [Figure 10] 10 is a flowchart showing an example of signal processing by a processor of the computer shown in FIG. DETAILED DESCRIPTION OF THE INVENTION

[0016] Hereinafter, embodiments of the present invention will be described in detail with reference to the drawings. However, the components described in the following embodiments are merely examples and are not intended to limit the technical scope of the present invention.

[0017] 1. First embodiment A signal processing device 100 according to a first embodiment of the present invention will be described with reference to Fig. 1. The signal processing device 100 in Fig. 1 is a device that obtains an estimated value e1(k) of a first signal from a first mixture signal xP(k) in which a first signal and a second signal are mixed.

[0018] As shown in FIG. 1, the signal processing device 100 includes a first input unit 101, a second input unit 102, an adaptive filter 103, a subtraction unit 104, an estimation unit 106, and a coefficient update control unit 107.

[0019] Of these, the first input unit 101 receives a first mixed signal xP(k) in which the first signal and the second signal are mixed. The second input unit 102 receives a second mixed signal xR(k) in which the third signal and the fourth signal are mixed. The first signal and the third signal originate from the same signal source A and are correlated with each other. The second signal and the fourth signal originate from the same signal source B and are correlated with each other.

[0020] The subtractor 104 receives the first mixture signal xP(k) and an estimate n1(k) of the second signal mixed in the first mixture signal xP(k), and outputs an estimate e1(k) of the first signal. Then, the adaptive filter 103 performs filtering on the second mixture signal xR(k) using a coefficient 141 that is updated based on the estimate e1(k) of the first signal, in order to obtain the estimate n1(k) of the second signal.

[0021] The estimation unit 106 estimates the ratio of the amplitudes or powers of the first and second signals as a first mixture ratio R1(k) using the estimated value e1(k) of the first signal, the estimated value n1(k) of the second signal, the first mixture signal xP(k), the second mixture signal xR(k), and the coefficient vector w1(k) of the adaptive filter 103. When the value of the first mixture ratio R1(k) obtained by the estimation unit 106 is large, the coefficient update control unit 107 outputs a control signal μ1(k) to the adaptive filter 103 to reduce the amount of update of the coefficients 141 of the adaptive filter 103.

[0022] According to this embodiment having such a configuration, the second signal can be removed from a mixed signal in which the first signal and the second signal are mixed, with a small amount of calculation and without delay. As a result, even if the gain of the acoustic impulse response approximated by the adaptive filter is less than 1, it is possible to obtain an estimate of the first signal with little residual second signal and little distortion.

[0023] [2, Second embodiment] As a signal processing device according to a second embodiment of the present invention, a noise canceller will be described which receives a noisy signal (a signal obtained by mixing a desired signal and noise) and a reference signal (mainly including a signal correlated with noise), cancels part or all of the noise, and outputs an emphasized signal (a signal obtained by emphasizing the desired signal). Here, the noisy signal corresponds to a first mixed signal obtained by mixing a first signal and a second signal, the reference signal corresponds to a second mixed signal obtained by mixing a third signal and a fourth signal, and the emphasized signal (an estimated value of the first signal) corresponds to the desired signal.

[0024] [2.1. Explanation of basic noise cancellation techniques] The following is a brief explanation of the basic technology of noise cancellation, which uses an adaptive filter to cancel noise, interference signals, echoes, etc. mixed in with a desired signal input from a microphone, handset, communication channel, etc., or to emphasize the desired signal.

[0025] As disclosed in Patent Documents 1 to 3, a two-input noise canceller generates, from a reference signal, a pseudo-noise (an estimated value of a second signal) corresponding to a noise component mixed into speech at the speech input terminal, using an adaptive filter that approximates the impulse response of the acoustic path from the noise source to the speech input terminal. The noise component is then suppressed by subtracting this pseudo-noise from a signal (first mixed signal) input to the speech input terminal. Here, the mixed signal refers to a signal in which a desired (speech) signal and noise are mixed, and is generally supplied to the speech input terminal from a microphone or handset. The reference signal is a signal correlated with the noise component in the noise source and is captured near the noise source. By capturing the reference signal near the noise source in this way, the reference signal can be considered to be approximately equal to the noise component in the noise source. The adaptive filter receives the reference signal supplied to the reference input terminal.

[0026] The coefficients of the adaptive filter are corrected by calculating the correlation between the error obtained by subtracting pseudo-noise from the noisy signal and the reference signal input to the reference input terminal. As coefficient correction algorithms for such adaptive filters, Patent Documents 1 to 3 disclose the "Least Mean-Square Algorithm (LMS)" and the "Learning Identification Method (LIM)." The LIM is also called the Normalized LMS (NLMS) algorithm.

[0027] Coefficient update using the normalized LMS algorithm is expressed by equation (1), where μ1(k) is the step size at time k. The coefficient vector w1(k) is defined by equation (2) using its elements. The reference signal vector xR(k) is expressed by equation (3) using its elements. Note that T represents the transpose of the vector, and the total number of coefficients is L.

number

[0028] The LMS algorithm and LIM are types of algorithms known as gradient methods, and the speed and accuracy of coefficient updates depend on a constant called the coefficient update step size. Filter coefficients are updated by multiplying the coefficient update step size by an error. To reduce interference with the coefficient update caused by the desired signal (the estimated value of the first signal) contained in the error, the coefficient update step size must be set to an extremely small value or zero. Constantly setting the coefficient update step size to a small value reduces the adaptive filter coefficients' ability to track environmental changes. Patent Documents 1 to 3 above disclose a method for solving the problem of increased output error or distortion of the desired signal. Furthermore, because the desired signal is generally speech, the term "speech" will be used hereinafter. However, the scope of the present invention is not limited to speech and encompasses any type of signal, including acoustic (audio) signals. Furthermore, the coefficient update algorithm is not limited to LMS or LIM.

[0029] [2.2. Configuration of noise canceller] 2 is a block diagram showing the overall configuration of a noise cancellation device 200 according to this embodiment. The noise cancellation device 200 may function as part of a device such as a digital camera, a laptop computer, a mobile phone, a hearing aid, a television, a smart speaker, or a robot, but the present invention is not limited to these and may be applied to any signal processing device that is required to cancel noise from an input signal.

[0030] As shown in Fig. 2, noise cancellation device 200 receives a noisy signal (first mixture signal) xP(k) containing a mixture of speech (first signal) and noise (second signal) from input terminal 201. It also receives a reference signal (second mixture signal) xR(k) containing a mixture of speech and noise from input terminal 202, and outputs an estimated value e1(k) of speech (enhanced signal) from output terminal 205. Adaptive filter 203 includes adaptive filter 103 and coefficient update control unit 107 shown in Fig. 1. Adaptive filter 203 receives first mixture ratio R1(k), calculates step size μ1(k), and updates coefficient vector w1(k) using the calculated step size μ1(k). The noise canceller 200 generates pseudo-noise n1(k) by modifying a reference signal xR(k) that is correlated with the noise to be cancelled using an adaptive filter 203, and then subtracts this from a noisy signal xP(k), which is a speech signal with noise superimposed thereon, thereby canceling the noise.

[0031] A noisy signal xP(k) is supplied to an input terminal 201 as a sequence of sample values. The noisy signal xP(k) is transmitted to a subtraction unit 204. A reference signal xR(k) is supplied to an input terminal 202 as a sequence of sample values. The reference signal xR(k) is transmitted to an adaptive filter 203 and an estimation unit 206.

[0032] The adaptive filter 203 performs a convolution operation on the reference signal xR(k) and the filter coefficients, and transmits the result as pseudo noise n1(k) to the subtraction unit 204 and the estimation unit 206. The adaptive filter 203 also supplies the coefficient vector w1(k) to the estimation unit 206.

[0033] The subtraction unit 204 receives the noisy signal xP(k) from the input terminal 201 and the pseudo-noise n1(k) from the adaptive filter 203. The subtraction unit 204 subtracts the pseudo-noise n1(k) from the noisy signal xP(k) and transmits the result as an estimated value e1(k) of the speech signal (estimated value of the first signal) to the output terminal 205, while simultaneously feeding it back to the adaptive filter 203.

[0034] Estimation unit 206 receives speech estimate e1(k), pseudo-noise n1(k) (output of adaptive filter 203), reference signal xR(k), noisy signal xP(k), and coefficient vector w1(k) of adaptive filter 203, estimates the ratio of the amplitude or power of speech to noise at input terminal 201 as first mixture ratio R1(k), and transmits this to adaptive filter 203. Adaptive filter 203 updates coefficient vector w1(k) using a small step size μ1(k) when first mixture ratio R1(k) is large and a large step size μ1(k) when first mixture ratio R1(k) is small. Methods for controlling the step size using the first mixture ratio R1(k), i.e., the estimated value of the signal-to-noise ratio, are disclosed in detail in Patent Documents 1 to 3. Furthermore, as disclosed in Patent Documents 1 to 3, first mixture ratio R1(k) may be averaged before being used to calculate step size μ1(k). The accuracy of the estimation of the ratio of the amplitude or power of speech to noise is improved.

[0035] 2.3. First Configuration of Estimation Unit 206 FIG. 3 is a block diagram showing a first internal configuration of estimation unit 206. Estimation unit 206 includes signal ratio estimation unit 301, signal ratio estimation unit 302, mixer 305, and correction unit 310. Signal ratio estimation unit 301 receives speech estimate e1(k) and pseudo-noise n1(k) and estimates the amplitude or power ratio of speech to noise at input terminal 201 as second mixture ratio R2(k). Second mixture ratio R2(k) may be the ratio of the amplitude or power of speech estimate e1(k) and pseudo-noise n1(k), or may be calculated by adding a small constant to the amplitude or power of the speech estimate e1(k) and pseudo-noise n1(k). Adding a small constant has the effect of preventing divergence of the quotient due to division. Alternatively, one or both of speech estimate e1(k) and pseudo-noise n1(k) may be averaged before use. Averaging can improve the accuracy of the ratio calculation.

[0036] The correction unit 310 receives the noisy signal xP(k) and the reference signal xR(k), corrects the reference signal xR(k), and obtains the corrected reference signal xRC(k). The powers of the noisy signal xP(k) and the reference signal xR(k) are averaged for M1 samples immediately after the noise canceller 200 starts operating, and the ratio is obtained as the scaling factor g1. The natural number M1 is a predetermined constant. It takes L sampling periods for non-zero input signal samples to be supplied to all coefficients of the adaptive filter 203, and normal operation begins. Therefore, one way to set M1 is to set M1 = L. The correction unit 310 obtains the corrected reference signal xRC(k) by multiplying the square root of the scaling factor g1 by the reference signal xR(k).

[0037] In general audio applications, the amplitude of the first signal can be assumed to be zero immediately after the device is turned on. This is because, for example, no conversation begins during the short period of time immediately after the device is turned on, such as 100 ms or 200 ms, and no voice (first signal) is present. However, background noise and the like are present immediately after the device is turned on. In this case, the noisy signal xP(k) is equal to the noise (second signal), so the scaling factor g1 can be estimated as the ratio of the power of the noisy signal (=noise) xP(k) to the power of the reference signal (mixed noise at input terminal 201) xR(k). The scaling factor g1 estimated in this way corresponds to the ratio of the input to the output when white noise is input to adaptive filter 203, i.e., the average gain of the impulse response of the acoustic system approximated by adaptive filter 203. The product of the power of reference signal xR(k) and scaling factor g1 is an approximation of the power of pseudonoise n1(k) (estimated value of the second signal) that reflects only the gain of adaptive filter 203. Therefore, the power of the pseudonoise n1(k) is more accurate than that of the reference signal xR(k). Therefore, by using the product of the power of the reference signal xR(k) and the scaling factor g1 instead of the power of the pseudo-noise n1(k), the ratio of the amplitude or power of the first signal to the second signal, i.e., the ratio of the speech and noise at the input terminal 201, can be calculated with higher accuracy.

[0038] The magnification factor g1 may be determined at any timing and any number of times as long as the amplitude of the audio (first signal) is 0. By determining the value of the magnification factor g1 more frequently, adaptive filter 203 can more accurately track changes in the impulse response of the acoustic system it approximates. The audio amplitude being 0 at times other than immediately after startup can be determined using the audio estimate value e1(k). Since the audio estimate value e1(k) always contains a certain error even when adaptive filter 203 has sufficiently converged, this error is taken into consideration when comparing audio estimate value e1(k) with a predetermined threshold value β1. Setting threshold β1 to a large value increases the frequency at which the amplitude is determined to be 0, but also increases the number of determination errors. Setting threshold β1 to a small value decreases the frequency at which the amplitude is determined to be 0, resulting in poor tracking of changes in the acoustic system impulse response.

[0039] Signal ratio estimation section 302 receives speech estimate e1(k) and corrected reference signal xRC(k), and estimates the ratio of the amplitude or power of speech to noise at input terminal 201 as third mixture ratio R3(k). Third mixture ratio R3(k) may be the ratio of the amplitude or power of speech estimate e1(k) to corrected reference signal xRC(k), or may be calculated by adding a small constant to these amplitudes or powers. Adding a small constant has the effect of preventing divergence of the quotient due to division. Alternatively, either or both of speech estimate e1(k) and corrected reference signal xRC(k) may be averaged before use.

[0040] Mixing unit 305 mixes second mixture ratio R2(k) and third mixture ratio R3(k) and outputs the mixture result as first mixture ratio R1(k). Second mixture ratio R2(k) and third mixture ratio R3(k) may be mixed by weighted addition, or may be mixed using a more complex higher-order polynomial. Prior to mixing, one or both of second mixture ratio R2(k) and third mixture ratio R3(k) may be averaged. Averaging can improve the calculation accuracy of first mixture ratio R1(k), i.e., the approximation accuracy of the amplitude or power of speech and noise at input terminal 201.

[0041] For simplicity, consider the case where the first mixture ratio R1(k) is calculated by mixing the second mixture ratio R2(k) and the third mixture ratio R3(k) using weighted addition. The weights for both are set to equal 1. The coefficients of the adaptive filter 203 are generally initialized to zero. Therefore, when coefficient updating begins, the pseudo-noise n1(k) is zero, and the second mixture ratio R2(k) has a zero denominator and is infinite. Therefore, when the step size μ1(k) of the adaptive filter 203 is calculated using the second mixture ratio R2(k), it becomes an extremely small value or zero, and the coefficient does not grow. If the coefficient does not grow, the pseudo-noise n1(k) does not increase, and the same problem persists.

[0042] On the other hand, the denominator of the third mixture ratio R3(k) is the corrected reference signal xRC(k), which is not necessarily zero at the start of coefficient updating. This is because the microphone input contains minute signals such as environmental noise. Even if the corrected reference signal xRC(k) is zero, it will not remain at zero. Therefore, the third mixture ratio R3(k) will not become infinite, and the corresponding step size μ1(k) will not become a minimum value. Therefore, the coefficients of the adaptive filter 203 grow as the coefficients are updated and converge to a value representing the acoustic characteristics of the path from the noise signal source to the input terminal 201. When the corrected reference signal xRC(k) is zero, the reference signal xR(k) will also be zero, and the coefficients of the adaptive filter 203 will not be updated. Therefore, even if the third mixture ratio R3(k) is an extremely large value, it does not cause any problems. However, when the coefficients of adaptive filter 203 grow to a certain extent and the pseudo-noise n1(k) grows, the third mixture ratio R3(k) has a lower approximation accuracy to the ratio of the amplitude or power of speech to noise at input terminal 201 than the second mixture ratio R2(k).

[0043] Therefore, mixer 305 sets the weight of third mixture ratio R3(k) to a large value when coefficient update of adaptive filter 203 starts, and decreases it as the coefficient grows. The weight of second mixture ratio R2(k) is set to a small value when coefficient update of adaptive filter 203 starts, and increases it over time. This means that the proportion of third mixture ratio R3(k) in first mixture ratio R1(k) decreases in accordance with the number of coefficient updates.

[0044] For example, if the weight of the third mixture ratio R3(k) is set to 1 when the coefficient update of adaptive filter 203 begins, the weight of the second mixture ratio R2(k) becomes 0 because the sum of the weights is 1. The growth of the coefficients corresponds to the number of coefficient updates. Therefore, the weight of the third mixture ratio R3(k) is set to 1 when the coefficient update of adaptive filter 203 begins, and the weight decreases toward 0 corresponding to the number of coefficient updates of coefficient vector w1(k). Correspondingly, the weight of the second mixture ratio R2(k) increases from 0 to 1. If the initial value of the weight of the third mixture ratio R3(k) is set to 1, and the weight becomes 1 at some point or is set to 1, the third mixture ratio R3(k) is switched to the second mixture ratio R2(k). Similar effects can be obtained even if the sum of the weights is set to a value other than 1 or if the initial value of the weight for the third mixture ratio R3(k) is set to a value other than 1. The weight can be determined using the amount of change in the coefficient vector w1(k) of the adaptive filter 203.

[0045] FIG. 4 is a diagram schematically illustrating the time transition of a signal according to the second embodiment, showing the corrected reference signal xRC(k), the pseudo-noise n1(k) output by the adaptive filter 203, and the coefficient vector w1(k). The vertical axis in FIG. 4 represents the signal value in terms of power, and the coefficient is the norm of the coefficient vector. The horizontal axis represents the number of coefficient updates of the adaptive filter 203, expressed as the number of samples k. Comparing xRC(k) and n1(k) is equivalent to comparing the third mixture ratio R3(k) and the second mixture ratio R2(k), which have the same numerator and the denominators xRC(k) and n1(k), respectively. Because xRC(k) is unrelated to the coefficient updates, n1(k) determines the relationship between the third mixture ratio R3(k) and the second mixture ratio R2(k). n1(k) is determined by xRC(k) and the coefficient vector w1(k), and its amplitude or power increases with the coefficient update, excluding changes due to increases or decreases in xRC(k). That is, xRC(k) is constant, and n1(k) increases with the increase in the coefficient vector w1(k) due to the coefficient update.

[0046] The relationship between the corrected reference signal xRC(k), the pseudo-noise n1(k) output by the adaptive filter 203, and the coefficient vector w1(k) described above is clear from FIG. 4. In FIG. 4, xRC(k), which is unrelated to the coefficient update, is shown as a constant for simplicity. n1(k), which depends on xRC(k), increases smoothly, and when the coefficient vector w1(k) converges, n1(k) also saturates. Because the amount of change in the coefficient vector w1(k) of the adaptive filter 203 decreases with coefficient update, the amount of change in the coefficient vector w1(k) can be used as an indicator of how much n1(k) has grown from zero. That is, the mixer 305 determines the weight of the third mixing ratio R3(k) based on the time change in the coefficient vector w1(k).

[0047] By correcting xR(k) by the scaling factor g1, the difference between the power of n1(k) and the power of xR(k) in FIG. 4 can be reduced. This means that the difference between the power of n1(k) and the power of xRC(k) is smaller than the difference between the power of n1(k) and the power of xR(k). In other words, xRC(k) can approximate n1(k) with higher accuracy than xR(k). This leads to more accurate step size control through the high-accuracy third mixture ratio R3(k), which is desirable for controlling adaptive filter 203. Therefore, it is possible to obtain an estimated speech value with less residual noise and less distortion.

[0048] The time change of the coefficient vector w1(k) may be the time change of the sum of squares or the sum of absolute values ​​of the elements of the coefficient vector w1(k), or the time change of the partial sum of squares or the partial sum of absolute values. When using the partial sum, the elements of the coefficient vector may be thinned out, or a portion of the coefficient vector may be cut out. By using the partial sum, it is possible to reduce the amount of calculation required for the time change of the coefficient while suppressing a decrease in the evaluation accuracy of the time change. Furthermore, the coefficient changes smoothly because it does not depend on the input signal. Therefore, it has smaller fluctuations than other indices that depend on the input signal, and can accurately detect a saturation state.

[0049] Equation (4) shows an example of mixing the second mixture ratio R2(k) and the third mixture ratio R3(k) by weighted addition. ζ1(k) expressed in equation (5) satisfies ζ1(0) = 0 and increases as the coefficients are updated. δ1(k), which corresponds to the time change in the coefficient vector w1(k) of the adaptive filter 203, can be calculated using equation (6). M is a predetermined natural number that determines the frequency at which the time change in the coefficient vector w1(k) is calculated. The larger M, the smaller the fluctuation in δ1(k) and the more accurate the detection of coefficient saturation. However, the larger the delay in the change in δ1(k), the longer the detection of coefficient saturation. Equation (5) represents an example in which, when the value of the time change δ1(k) becomes less than the threshold ε1, the content ratio of the third mixture ratio R3(k) in the first mixture ratio R1(k) is set to 0, and the first mixture ratio R1(k) is switched from the third mixture ratio R3(k) to the second mixture ratio R2(k). This is an example of a method for determining whether the value of the time change δ1(k) has become sufficiently small. If ζ1(k) = δ1(k) / |δ1(M)| is used instead of equation (5), the content ratio of the third mixture ratio R3(k) is determined by the time change of the coefficient vector w1(k).

number

[0050] In this way, the mixer 305 can set the content ratio of the third mixture ratio R3(k) in the first mixture ratio R1(k) to either 100% or 0% based on the comparison result between the time change δ1(k) and the threshold ε1. Alternatively, the mixer 305 may repeatedly compare the time change δ1(k) of the coefficient vector w1(k) with the threshold ε1, and set the content ratio of the third mixture ratio R3(k) in the first mixture ratio R1(k) to 100% if the time change δ1(k) of the coefficient vector w1(k) is equal to or greater than the threshold ε1. Otherwise, the mixer 305 may repeatedly set the content ratio of the third mixture ratio R3(k) in the first mixture ratio R1(k). This allows the first mixture ratio R1(k) to have high resilience to environmental changes, enabling more accurate coefficient update even when, for example, the amount of update of the coefficient vector w1(k) of the adaptive filter 203 increases rapidly.

[0051] The mixer 305 may set the content ratio of the third mixture ratio R3(k) in the first mixture ratio R1(k) to 0% if the time change δ1(k) of the coefficient vector w1(k) is less than the threshold ε1 for a predetermined first constant L1 or more consecutive times. Otherwise, the mixer 305 may set the content ratio of the third mixture ratio R3(k) in the first mixture ratio R1(k) to 100%. After the content ratio of the third mixture ratio R3(k) is set to 0% once, the comparison of δ1(k) and ε1 may be stopped by setting the threshold to a sufficiently large value. Here, L1 is an integer equal to or greater than 2.

[0052] The mixing unit 305 may set the content ratio of the third mixture ratio R3(k) in the first mixture ratio R1(k) to 0% when δ1(k) is less than the threshold ε1 for a predetermined third constant L3 or more consecutive times in a section where the time change δ1(k) of the coefficient vector w1(k) has a number of samples equal to the predetermined second constant L2. Otherwise, the content ratio of the third mixture ratio R3(k) in the first mixture ratio R1(k) may be set to 100%. The second constant L2 and the third constant L3 are both integers greater than or equal to 2, satisfying L2 > L3. Furthermore, the continuous condition may be removed from the above evaluation, and the number of times δ1(k) is less than the threshold ε1 for a predetermined third constant L3 or more may be evaluated. After the content ratio of the third mixture ratio R3(k) is set to 0% once, the evaluation of δ1(k) may be stopped by setting L3 to a value greater than L2.

[0053] Note that mixer 305 may set the minimum value of the weights of second mixture ratio R2(k) and third mixture ratio R3(k) to a value greater than 0 rather than 0.

[0054] 2.4. Second Configuration of Estimation Unit 206 FIG. 5 is a block diagram showing a second internal configuration of the estimation unit 206. The estimation unit 206 includes a correction unit 310, a mixer 506, and a signal ratio estimation unit 503. The correction unit 310 receives the noisy signal xP(k) and the reference signal xR(k), corrects the reference signal xR(k), and obtains a corrected reference signal xRC(k). The operation of the correction unit 310 is similar to that of the correction unit 310 already described. The mixer 506 mixes the corrected reference signal xRC(k) (corrected second mixed signal) with the pseudo-noise n1(k) (estimated value of the second signal) based on the coefficient vector w1(k) to generate a first mixed signal n3(k). The signal ratio estimation unit 503 receives the estimated value e1(k) of speech and the first mixed signal n3(k), and estimates the amplitude or power ratio of speech to noise at the input terminal 201 as a first mixed ratio R1(k). The first mixture ratio R1(k) may be the ratio of the amplitude or power of the speech estimate e1(k) to the first mixture signal n3(k), or may be calculated by adding a small constant to the amplitude or power. Adding a small constant has the effect of preventing the quotient from diverging due to division. Alternatively, either or both of the estimate e1(k) and the first mixture signal n3(k) may be averaged before use. Averaging can improve the accuracy of the ratio calculation.

[0055] The second internal configuration of estimation unit 206 shown in Fig. 5 is equivalent to the first internal configuration of estimation unit 206 shown in Fig. 3. That is, in the first internal configuration shown in Fig. 3, two estimates for the amplitude or power ratio of speech to noise at input terminal 201 are generated in signal ratio estimation units 301 and 302, and then these estimates are mixed to calculate first mixture ratio R1(k). The second internal configuration of estimation unit 206 shown in Fig. 5 mixes two types of noise estimates, i.e., corrected reference signal xRC(k) and pseudo-noise n1(k), to generate first mixed signal n3(k), determine the denominator, and calculate first mixture ratio R1(k) by combining this with the numerator, speech estimate e1(k). These two types of configurations are possible because the same numerator, i.e., speech estimate e1(k), is used when estimating the amplitude or power ratio of speech to noise at input terminal 201 in the first internal configuration shown in Fig. 3 and the second internal configuration shown in Fig. 5. The second internal configuration of the estimation unit 206 shown in FIG. 5 has fewer components and is simpler than the first internal configuration shown in FIG.

[0056] The operation of mixer 506 is similar to that of mixer 305, except that the second and third mixing ratios, which are input signals, are replaced with pseudo-noise and a reference signal (second mixing signal), respectively. Therefore, the description of the operation of mixer 305 can be similarly applied to mixer 506.

[0057] With the above configuration, this embodiment can smoothly update the coefficients without forcibly setting a special value for the step size, even if the gain of the acoustic impulse response approximated by the adaptive filter 203 is less than 1. As a result, it is possible to obtain an output signal with little residual noise and little signal distortion.

[0058] 3. Third Embodiment In the explanation so far, it has been assumed that the reference signal is noise itself, by capturing the reference signal near the noise source. However, in reality, there are cases where this condition cannot be met. In such cases, the reference signal is composed of noise and an audio signal mixed in with it. Such audio signal components mixed into the reference signal are called crosstalk. Patent Document 3 discloses the configuration of a noise cancellation device for use when crosstalk exists.

[0059] In this embodiment, a second adaptive filter is introduced to cancel crosstalk, similar to the noise cancellation. The second adaptive filter, which approximates the impulse response of the acoustic path (crosstalk path) from the audio signal source to the reference input terminal, is used to generate pseudo crosstalk corresponding to the audio signal components mixed in at the reference input terminal. Then, by subtracting this pseudo crosstalk from the signal (reference signal) input to the reference input terminal, the audio signal components (crosstalk) mixed in the reference input are canceled.

[0060] A noise cancellation device according to a third embodiment of the present invention will be described with reference to Fig. 6. Compared to the second embodiment, the noise cancellation device according to this embodiment includes a subtraction unit 804 and an adaptive filter 803 in addition to the subtraction unit 204 and the adaptive filter 203, and the estimation unit 206 is replaced with an estimation unit 806. The other configurations and operations are the same as those of the second embodiment, so the same components are denoted by the same reference numerals and detailed description will be omitted.

[0061] Noise cancellation device 800 uses an adaptive filter to modify a signal (output at output terminal 205 = estimated speech signal or emphasis signal) correlated with the crosstalk (third signal) to be cancelled, to generate pseudo crosstalk n2(k) (an estimated value of the third signal). Then, this is subtracted from a reference signal xR(k) containing a mixture of speech and noise, thereby canceling the crosstalk. When updating the coefficients of adaptive filter 803, the step size is controlled using a fourth mixture ratio R4(k) that approximates the ratio of the amplitude or power of the fourth signal to the third signal. This allows for smooth coefficient updating, resulting in an output signal with minimal residual noise and minimal signal distortion.

[0062] The noisy signal xP(k) is supplied to the input terminal 201 as a sample value sequence and transmitted to a subtraction unit 204. The reference signal xR(k) is supplied to the input terminal 202 as a sample value sequence and transmitted to a subtraction unit 804.

[0063] The subtraction unit 804 receives the reference signal xR(k) from the input terminal 202 and the pseudo crosstalk n2(k) from the adaptive filter 803. The subtraction unit 804 subtracts the pseudo crosstalk n2(k) from the reference signal xR(k) and transmits the result as a noise estimate e2(k) (estimate of the fourth signal) to the output terminal 805, while simultaneously feeding it back to the adaptive filter 803. The subtraction unit 804 also supplies the noise estimate e2(k) to the estimation unit 806.

[0064] The adaptive filter 803 performs a convolution operation on the speech estimate e1(k) (emphasis signal) and the filter coefficients, and transmits the result as pseudo crosstalk n2(k) (estimate of the third signal) to the subtraction unit 804 and the estimation unit 806. The adaptive filter 803 also supplies the coefficient vector w2(k) to the estimation unit 806.

[0065] Estimation unit 806 receives speech estimate e1(k), pseudo-noise n1(k) (output of adaptive filter 203), noise estimate e2(k) (estimate of the fourth signal), coefficient vector w1(k) of adaptive filter 203, and noisy signal xP(k), estimates the ratio of the amplitude or power of speech to noise at input terminal 201 as first mixture ratio R1(k), and transmits this to adaptive filter 203. Adaptive filter 203 updates its coefficients using a small step size μ1(k) when first mixture ratio R1(k) is large and a large step size μ1(k) when first mixture ratio R1(k) is small. Note that adaptive filter 203 shown in FIG. 6 differs from adaptive filter 203 shown in FIG. 2 in that noise estimate e2(k) is input instead of reference signal xR(k). Methods for controlling the step size using the first mixture ratio R1(k), i.e., an estimated value of the signal-to-noise ratio, are disclosed in detail in Patent Documents 1 to 3. As disclosed in Patent Documents 1 to 3, the first mixture ratio R1(k) may be averaged and then used to calculate the step size μ1(k). This improves the estimation accuracy of the ratio of the amplitude or power of speech to noise at input terminal 201.

[0066] Estimation unit 806 further receives pseudo crosstalk n2(k) (output of adaptive filter 803), coefficient vector w2(k) of adaptive filter 803, and reference signal xR(k), estimates the ratio of the amplitude or power of noise and crosstalk (fourth signal and third signal) at input terminal 202 as fourth mixing ratio R4(k), and transmits this to adaptive filter 803. The size of coefficient vector w2(k) shown in equation (7) may be equal to or different from the size L of coefficient vector w1(k).

number

[0067] Adaptive filter 803 updates the coefficients using a small step size μ2(k) when fourth mixing ratio R4(k) is large and a large step size μ2(k) when fourth mixing ratio R4(k) is small. Methods for controlling the step size μ2(k) using the estimated value of fourth mixing ratio R4(k), i.e., the noise-to-crosstalk ratio at input terminal 202, are disclosed in detail in Patent Documents 1 to 3. Furthermore, as disclosed in detail in Patent Documents 1 to 3, fourth mixing ratio R4(k) may be averaged before being used to calculate step size μ2(k). This improves the estimation accuracy of the ratio of the amplitude or power of noise to crosstalk at input terminal 202.

[0068] 3.1. First Configuration of Estimation Unit 806 FIG. 7 is a block diagram showing a first internal configuration of estimating section 806. In addition to the configuration of estimating section 206, estimating section 806 includes signal ratio estimating section 901, signal ratio estimating section 902, mixing section 905, and correcting section 710. Correcting section 310 shown in FIG. 7 differs from correcting section 310 shown in FIG. 3 in the following respects. First, instead of reference signal xR(k), a noise estimate (fourth signal estimate) e2(k) is input. Second, instead of corrected reference signal xRC(k), a corrected noise estimate e2C(k) is output. The remaining operations are the same. Signal ratio estimating section 302 shown in FIG. 7 is the same as signal ratio estimating section 302 shown in FIG. 3 except that a corrected noise estimate e2C(k) is used instead of corrected reference signal xRC(k).

[0069] The signal ratio estimator 901 receives the noise estimate e2(k) and the pseudo crosstalk n2(k) and estimates the ratio of the amplitude or power of the noise to the crosstalk as a fifth mixture ratio R5(k). The fifth mixture ratio R5(k) may be the ratio of the amplitude or power of the noise estimate e2(k) to the pseudo crosstalk n2(k), or may be calculated by adding a small constant to the amplitude or power. Adding a small constant has the effect of preventing divergence of the quotient due to division. Alternatively, one or both of the noise estimate e2(k) and the pseudo crosstalk n2(k) may be averaged before use. Averaging can improve the accuracy of the ratio calculation.

[0070] Correction unit 710 receives reference signal xR(k) and speech estimate e1(k), corrects speech estimate e1(k), and obtains corrected speech (first signal) estimate e1C(k). The operation of correction unit 710 is the same as that of correction unit 310, except that noise estimate e2(k) is replaced with speech estimate e1(k) and noisy signal xP(k) is replaced with reference signal xR(k). However, there is a difference in how the scaling factor g2 used for correction is obtained.

[0071] Similar to the correction unit 310, the correction unit 710 can average the power of the reference signal xR(k) and the speech estimate e1(k) for M2 samples immediately after the noise cancellation device 800 starts operating and calculate the ratio as the magnification g2. The natural number M2 is a predetermined constant, and similar to the correction unit 310, M2 can be set to the number of samples equal to the number of taps of the adaptive filter 803. When the noise is continuous and does not have silent intervals, an appropriate magnification g2 cannot be calculated using this method, so g2 = g1 can be used. This is because the magnifications g1 and g2 correspond to the same acoustic space, and the values ​​of g1 and g2, which are dominated by the reflectance of the acoustic space, can be considered approximately equal. By using the product of the magnification g2 calculated in this way and the power of the speech estimate e1(k) instead of the power of the pseudo crosstalk n2(k), the ratio of the amplitude or power of the fourth signal and the third signal, i.e., the noise and crosstalk at the input terminal 202, can be calculated with higher accuracy.

[0072] The magnification factor g2 may be determined at any timing and any number of times as long as the amplitude of the noise (fourth signal) is 0. By determining the value of the magnification factor g2 more frequently, adaptive filter 803 can more accurately track changes in the impulse response of the acoustic system it approximates. The noise estimate e2(k) can be used to determine that the noise amplitude is 0 other than immediately after startup. Since the noise estimate e2(k) always contains a certain error even when adaptive filter 803 has sufficiently converged, this error is taken into consideration when comparing the noise estimate e2(k) with a predetermined threshold β2. Setting threshold β2 to a large value increases the frequency at which the amplitude is determined to be 0, but also increases the likelihood of incorrect determination. Setting threshold β2 to a small value decreases the frequency at which the amplitude is determined to be 0, resulting in poor tracking of changes in the acoustic system impulse response.

[0073] Signal ratio estimator 902 receives noise estimate e2(k) and corrected speech estimate e1C(k) (corrected first signal estimate) and estimates the ratio of the amplitude or power of the noise to the crosstalk at input terminal 202 as sixth mixture ratio R6(k). Sixth mixture ratio R6(k) may be the ratio of the amplitude or power of noise estimate e2(k) to corrected speech estimate e1C(k), or may be calculated by adding a small constant to these amplitudes or powers. Adding a small constant has the effect of preventing divergence of the quotient due to division. Alternatively, either or both of noise estimate e2(k) and corrected speech estimate e1C(k) may be averaged before use.

[0074] Mixing unit 905 mixes fifth mixture ratio R5(k) and sixth mixture ratio R6(k) using coefficient vector w2(k) of adaptive filter 803 and outputs the mixed result as fourth mixture ratio R4(k). The fifth mixture ratio R5(k) and sixth mixture ratio R6(k) may be mixed by weighted addition using coefficient vector w2(k) of adaptive filter 803, or may be mixed using a more complex higher-order polynomial. Prior to mixing, either or both of fifth mixture ratio R5(k) and sixth mixture ratio R6(k) may be averaged. Averaging can improve the calculation accuracy of fourth mixture ratio R4(k), i.e., the approximation accuracy of the amplitude or power of noise and crosstalk.

[0075] The operation of mixer 905 becomes the same as that of mixer 305 by changing the input and output signals of mixer 305 to those of mixer 905 as shown in FIG. 7, so a detailed description will be omitted.

[0076] 3.3. Second Configuration of Estimation Unit 806 FIG. 8 is a block diagram showing a second internal configuration of the estimation unit 806. In addition to the mixer 506, the signal ratio estimation unit 503, and the correction unit 310, the estimation unit 806 further includes a mixer 1106, a signal ratio estimation unit 1103, and a correction unit 710. The correction units 310 and 710 in FIG. 8 are identical to the correction units 310 and 710 in FIG. 7 in terms of input / output signals and operation, and therefore detailed description thereof will be omitted. The mixer 506 in FIG. 8 is configured to input a corrected noise estimate value e2C(k) instead of the corrected reference signal xRC(k) (second mixed signal) in the mixer 506 in FIG. 5. The operation of the mixer 506 in FIG. 8 is identical to that of the mixer 506 in FIG. 5, and therefore detailed description thereof will be omitted.

[0077] The mixer 1106 mixes the corrected speech estimate e1C(k) (the corrected emphasis signal, i.e., the corrected estimate of the first signal) and the pseudo crosstalk n2(k) (the estimate of the third signal) using the coefficient vector w2(k) of the adaptive filter 803 to generate a second mixed signal n4(k). The operation of the mixer 1106 becomes the same as that of the mixer 506 by changing the input and output signals of the mixer 506 to those of the mixer 1106 as shown in Fig. 8, and therefore a detailed description thereof will be omitted.

[0078] The signal ratio estimator 1103 receives the noise estimate e2(k) and the second mixed signal n4(k) and estimates the ratio of the amplitude or power of the noise to the crosstalk as a fourth mixed ratio R4(k). The fourth mixed ratio R4(k) may be the ratio of the amplitude or power of the noise estimate e2(k) to the second mixed signal n4(k), or may be calculated by adding a small constant to the amplitude or power of the noise estimate e2(k) and the second mixed signal n4(k). Adding a small constant has the effect of preventing divergence of the quotient due to division. Alternatively, one or both of the noise estimate e2(k) and the second mixed signal n4(k) may be averaged before use. Averaging can improve the accuracy of the ratio calculation.

[0079] The second internal configuration of estimator 806 shown in Fig. 8 is equivalent to the first internal configuration of estimator 806 shown in Fig. 7. That is, in the first internal configuration shown in Fig. 7, two estimates for the ratio of the amplitude or power of noise to crosstalk are generated by signal ratio estimators 901 and 902, and then these estimates are mixed to calculate fourth mixture ratio R4(k). In the second internal configuration shown in Fig. 8, two types of estimates, i.e., corrected speech estimate e1C(k) and pseudo crosstalk n2(k), are mixed using coefficient vector w2(k) of adaptive filter 803 to generate second mixed signal n4(k), determine the denominator, and then calculate fourth mixture ratio R4(k) by combining this with noise estimate e2(k), which is the numerator. These two types of configurations are possible because the first internal configuration shown in Fig. 7 and the second internal configuration shown in Fig. 8 use the same numerator, i.e., noise estimate e2(k), when estimating the ratio of the amplitude or power of noise to crosstalk at input terminal 202. The second internal configuration of the estimating unit 806 shown in FIG. 8 has fewer components and is simpler than the first internal configuration shown in FIG.

[0080] With the above configuration, this embodiment can smoothly update the coefficients without forcibly setting a special value for the step size, even when the gain of the acoustic impulse response that the adaptive filters 203 and 803 approximate is less than 1 and crosstalk exists. As a result, it is possible to obtain an output signal with little residual noise and little signal distortion.

[0081] 4. Other Embodiments Although multiple embodiments of the present invention have been described in detail above, any system or device that combines the separate features included in each embodiment also falls within the scope of the present invention.

[0082] The present invention may be applied to a system consisting of multiple devices, or to a single device. Furthermore, the present invention is also applicable when an information processing program (signal processing program) that realizes the functions of the above-described embodiments is supplied directly or remotely to a system or device. Such a program is executed by a processor such as a DSP (Digital Signal Processor) that constitutes a signal processing device or noise cancellation device. Furthermore, the scope of the present invention also includes a program installed on a computer to realize the functions of the present invention, a medium storing the program, and a WWW (World Wide Web) server from which the program can be downloaded.

[0083] 9 is a configuration diagram of a computer 1200 that executes a signal processing program when the first to third embodiments are configured by the signal processing program. The computer 1200 includes an input unit 1201, a processor 1203, an output unit 1202, and a memory 1204.

[0084] The processor 1203 controls the operation of the computer 1200 by reading a signal processing program stored in the memory 1204. The processor 1203 is, for example, a processor such as a DSP, a CPU (Central Processing Unit), or an MPU (Micro-Processing Unit). The memory 1204 includes one or more of a RAM (Random Access Memory), a ROM (Read-Only Memory), a flash memory, an EPROM (Erasable Programmable Read-Only Memory), and an EEPROM (Electrically Erasable Programmable Read-Only Memory).

[0085] Fig. 10 is a flowchart showing an example of signal processing by the processor of the computer shown in Fig. 9. The example shown in Fig. 10 is a flowchart in the case where the computer 1200 functions as the signal processing device 100 according to the first embodiment.

[0086] As shown in FIG. 10, in step S10, the processor 1203 executing the signal processing program first inputs, from the input unit 1201, a first mixture signal xP(k) in which a first signal and a second signal are mixed, and then inputs a second mixture signal xR(k) in which a third signal correlated with the first signal and a fourth signal correlated with the second signal are mixed. In step S11, the processor 1203 processes the second mixture signal xR(k) with a first adaptive filter (adaptive filter 103) to generate an estimate n1(k) of the second signal, and in step S12, generates an estimate e1(k) of the first signal from the first mixture signal xP(k) and the estimate n1(k) of the second signal. In step S13, the processor 1203 estimates the ratio of the amplitudes or powers of the first signal and the second signal as a first mixture ratio R1(k) using the estimated value e1(k) of the first signal, the estimated value n1(k) of the second signal, the second mixture signal xR(k), the first mixture signal xP(k), and the coefficient vector w1(k) of the first adaptive filter (adaptive filter 103). In step S14, the processor 1203 controls the generation of the estimated value n1(k) of the second signal using the first mixture ratio R1(k), thereby achieving the same effects as in the first embodiment.

[0087] Furthermore, when the computer 1200 functions as the signal processing device according to the second embodiment, the processor 1203 receives the first mixture signal xP(k) and the second mixture signal xR(k) in step S10, and processes the second mixture signal xR(k) using a first adaptive filter (adaptive filter 203) in step S11 to generate an estimated value n1(k) of the second signal. In step S12, the processor 1203 generates an estimated value e1(k) of the first signal from the first mixture signal xP(k) and the estimated value n1(k) of the second signal. In step S13, the processor 1203 generates a first mixture ratio R1(k) using the estimated value e1(k) of the first signal, the estimated value n1(k) of the second signal, the second mixture signal xR(k), the first mixture signal xP(k), and the coefficient 141 of the first adaptive filter (adaptive filter 203). In step S14, the processor 1203 controls the generation of the estimate n1(k) of the second signal using the first mixture ratio R1(k).

[0088] Furthermore, when the computer 1200 functions as the signal processing device according to the third embodiment, the processor 1203 generates a first mixture ratio R1(k) using the estimated value e1(k) of the first signal, the estimated value n1(k) of the second signal, the estimated value e2(k) of the fourth signal, the first mixture signal xP(k), and the coefficient 141 of the first adaptive filter (adaptive filter 203) in step S13. Furthermore, the processor 1203 processes the estimated value e1(k) of the first signal with the second adaptive filter (adaptive filter 803) to generate an estimated value n2(k) of the third signal, and subtracts the estimated value n2(k) of the third signal from the second mixture signal xR(k) to generate an estimated value e2(k) of the fourth signal. In addition, processor 1203 processes the estimated value e2(k) of the fourth signal instead of the estimated value xR(k) of the third signal in the first adaptive filter (adaptive filter 203), and further estimates the amplitude or power ratio of noise to crosstalk as a fourth mixture ratio R4(k) using the estimated value e2(k) of the fourth signal, the estimated value n2(k) of the third signal, the estimated value e1(k) of the first signal, the second mixture signal xR(k), and the coefficients of the second adaptive filter (adaptive filter 803).

[0089] [5. Other] Furthermore, among the processes described in the above embodiments, all or part of the processes described as being performed automatically can be performed manually, or all or part of the processes described as being performed manually can be performed automatically using known methods. In addition, the information including the processing procedures, specific names, various data, and parameters shown in the above documents and drawings can be changed as desired unless otherwise specified. For example, the various information shown in each drawing is not limited to the information shown in the drawings.

[0090] Furthermore, the components of each device shown in the figure are conceptual functional components and do not necessarily have to be physically configured as shown in the figure. In other words, the specific form of distribution and integration of each device is not limited to that shown in the figure, and all or part of them can be functionally or physically distributed and integrated in any unit depending on various loads, usage conditions, etc.

[0091] [6. Effects] As described above, the signal processing device 100 according to the first embodiment includes a first input unit 101 (corresponding to an example of a first input means), a second input unit 102 (corresponding to an example of a second input means), an adaptive filter 103 (corresponding to an example of a first adaptive filter), a subtraction unit 104 (corresponding to an example of a first subtraction unit), and an estimation unit 106. The first input unit 101 receives a first mixture signal xP(k) in which a first signal and a second signal are mixed. The second input unit 102 receives a second mixture signal xR(k) in which a third signal correlated with the first signal and a fourth signal correlated with the second signal are mixed. The adaptive filter 103 filters the second mixture signal xR(k) to generate an estimate n1(k) of the second signal. The subtractor 104 generates an estimate e1(k) of the first signal from the first mixture signal xP(k) and the estimate n1(k) of the second signal. The estimator 106 estimates a first mixture ratio R1(k), which is the ratio of the amplitudes or powers of the first signal and the second signal at the input terminal 201, using the estimate e1(k) of the first signal, the estimate n1(k) of the second signal, the second mixture signal xR(k), the first mixture signal xP(k), and the coefficients 141 of the adaptive filter 103. When the value of the first mixture ratio R1(k) obtained by the estimator 106 is large, the coefficient update controller 107 outputs a control signal μ1(k) to the adaptive filter 103 to reduce the amount of update of the coefficients 141 of the adaptive filter 103. The signal processing device 100 controls the adaptive filter 103 using the control signal μ1(k). As a result, the signal processing device 100 can achieve both fast convergence and a low-distortion output signal without manually controlling the step size μ1(k) even if the gain of the acoustic impulse response approximated by the adaptive filter is less than 1.

[0092] The signal processing device according to the second embodiment includes an input terminal 201 (corresponding to an example of a first input means), an input terminal 202 (corresponding to an example of a second input means), an adaptive filter 203 (corresponding to an example of a first adaptive filter), a subtraction unit 204 (corresponding to an example of a first subtraction unit), and an estimation unit 206. The input terminal 201 receives a noisy signal xP(k) (corresponding to an example of a first mixed signal) in which a speech signal (corresponding to an example of a first signal) and noise (corresponding to an example of a second signal) are mixed. The input terminal 202 receives a reference signal xR(k) (corresponding to an example of a second mixed signal) in which a signal correlated with the speech signal (corresponding to an example of a third signal) and a signal correlated with the noise (corresponding to an example of a fourth signal) are mixed. The adaptive filter 203 filters the reference signal xR(k) to generate pseudo-noise n1(k) (corresponding to an example of an estimated value of the second signal). The subtractor 104 generates a speech estimate e1(k) (corresponding to an example of the first signal estimate e1(k)) from the noisy signal xP(k) and the pseudo-noise n1(k). The estimator 206 estimates the ratio of the amplitude or power of the speech signal to the noise at the input terminal 201 as a first mixture ratio R1(k) using the speech estimate e1(k), the pseudo-noise n1(k), the reference signal xR(k), the noisy signal xP(k), and the coefficient vector w1(k) of the adaptive filter 203. The noise canceller 200 controls the adaptive filter 203 using the first mixture ratio R1(k). As a result, the signal processing apparatus according to the second embodiment can achieve both fast convergence and a low-distortion output signal without manually controlling the step size μ1(k), even if the gain of the acoustic impulse response approximated by the adaptive filter is less than 1.

[0093] Furthermore, estimation unit 206 includes signal ratio estimation unit 301 (corresponding to an example of a first signal ratio estimation unit), signal ratio estimation unit 302 (corresponding to an example of a second signal ratio estimation unit), correction unit 310, and mixer 305 (corresponding to an example of a first mixer). Signal ratio estimation unit 301 uses speech estimate value e1(k) and pseudo-noise n1(k) to estimate the amplitude or power ratio of the speech signal to noise at input terminal 201 as second mixture ratio R2(k). Correction unit 310 receives reference signal xR(k) and noisy signal xP(k) as input and calculates corrected reference signal xRC(k). Signal ratio estimation unit 302 uses speech estimate value e1(k) and corrected reference signal xRC(k) to estimate the amplitude or power ratio of the speech signal to noise at input terminal 201 as third mixture ratio R3(k). The mixer 305 generates the first mixture ratio R1(k) by mixing the second mixture ratio R2(k) and the third mixture ratio R3(k) based on the time change of the coefficient vector w1(k) of the adaptive filter 203. As a result, the signal processing device according to the second embodiment can achieve both fast convergence and a low-distortion output signal without manually controlling the step size μ1(k), even if the gain of the acoustic impulse response approximated by the adaptive filter is less than 1.

[0094] Furthermore, mixing unit 305 sets the content ratio of third mixing ratio R3(k) in first mixing ratio R1(k) to 100% when coefficient update of adaptive filter 203 starts, and sets the content ratio of third mixing ratio R3(k) in first mixing ratio R1(k) to 0% when the change over time of coefficient vector w1(k) of adaptive filter 203 becomes sufficiently small. As a result, the signal processing device according to the second embodiment can achieve both fast convergence and low-distortion output signals without manually controlling step size μ1(k), even if the gain of the acoustic impulse response approximated by the adaptive filter is less than 1.

[0095] The estimation unit 206 also includes a correction unit 310, a mixer 506 (corresponding to an example of a second mixer), and a signal ratio estimation unit 503 (corresponding to an example of a third signal ratio estimation unit). The correction unit 310 receives the reference signal xR(k) and the noisy signal xP(k) as input and calculates a corrected reference signal xRC(k). The mixer 506 mixes the corrected reference signal xRC(k) and a pseudo-noise n1(k) based on the time variation of the coefficient vector w1(k) of the adaptive filter 203 to generate a first mixed signal n3(k). The signal ratio estimation unit 503 uses the first mixed signal n3(k) and the estimated value e1(k) of the speech to estimate the ratio of the amplitude or power of the speech signal to noise at the input terminal 201 as a first mixed ratio R1(k). As a result, the signal processing device according to the second embodiment can achieve both fast convergence and low-distortion output signals without manually controlling the step size μ1(k), even if the gain of the acoustic impulse response approximated by the adaptive filter is less than 1.

[0096] Furthermore, the mixer 506 sets the content ratio of the corrected reference signal xRC(k) in the first mixed signal n3(k) to 100% when the coefficient update of the adaptive filter 203 starts, and sets the content ratio of the corrected reference signal xRC(k) in the first mixed signal n3(k) to 0% when the time change of the coefficient vector w1(k) of the adaptive filter 203 becomes sufficiently small. As a result, the signal processing device according to the second embodiment can achieve both fast convergence and a low-distortion output signal without manually controlling the step size μ1(k), even if the gain of the acoustic impulse response approximated by the adaptive filter is less than 1.

[0097] The signal processing device according to the third embodiment includes an input terminal 201, an input terminal 202, an adaptive filter 203, a subtraction unit 204, a subtraction unit 804 (corresponding to an example of a second subtraction unit), an adaptive filter 803 (corresponding to an example of a second adaptive filter), and an estimation unit 806. The adaptive filter 803 performs filtering on an estimated value e1(k) of a first signal to generate pseudo crosstalk n2(k) (corresponding to an example of an estimated value of a third signal). The subtraction unit 804 subtracts the pseudo crosstalk n2(k) from a reference signal xR(k) to generate an estimated value e2(k) of noise (corresponding to an example of an estimated value of a fourth signal). The adaptive filter 203 receives the estimated value e2(k) of noise instead of the reference signal xR(k). In addition to the functions of the estimation unit 206, the estimation unit 806 further uses the noise estimate e2(k), the pseudo crosstalk n2(k), the first signal estimate e1(k), the reference signal xR(k), and the coefficients of the adaptive filter 803 to estimate the ratio of the amplitude or power of the noise to the crosstalk at the input terminal 202 as a fourth mixing ratio R4(k). The signal processing device according to the third embodiment controls the adaptive filter 803 using the fourth mixing ratio R4(k). As a result, the signal processing device according to the third embodiment can achieve both fast convergence and a low-distortion output signal without manually controlling the step size μ2(k), even if the gain of the acoustic impulse response approximated by the adaptive filter is less than 1.

[0098] Furthermore, estimation unit 806 includes signal ratio estimation unit 301, correction unit 310, signal ratio estimation unit 302, mixer 305, signal ratio estimation unit 901 (corresponding to an example of a fourth signal ratio estimation unit), correction unit 710, signal ratio estimation unit 902 (corresponding to an example of a fifth signal ratio estimation unit), and mixer 905 (corresponding to an example of a third mixer). Signal ratio estimation unit 301 uses speech estimate value e1(k) and pseudo-noise n1(k) to estimate the ratio of amplitude or power of the speech signal to noise at input terminal 201 as second mixing ratio R2(k). Correction unit 310 receives noise estimate value e2(k) and noisy signal xP(k) as input and calculates corrected noise estimate value e2C(k). Signal ratio estimator 302 estimates the amplitude or power ratio of speech to noise at input terminal 201 as third mixture ratio R3(k) using speech estimate e1(k) and corrected noise estimate e2C(k). Mixer 305 mixes second mixture ratio R2(k) and third mixture ratio R3(k) based on time changes in coefficient vector w1(k) of adaptive filter 203 to generate first mixture ratio R1(k). Signal ratio estimator 901 estimates the amplitude or power ratio of noise to crosstalk at input terminal 202 as fifth mixture ratio R5(k) using noise estimate e2(k) and pseudo crosstalk n2(k). Corrector 710 receives speech estimate e1(k) and reference signal xR(k) as input and calculates corrected speech estimate e1C(k). The signal ratio estimator 902 uses the noise estimate e2(k) and the corrected speech estimate e1C(k) to estimate the ratio of the amplitude or power of the noise to the crosstalk at the input terminal 202 as a sixth mixture ratio R6(k). The mixer 905 mixes the fifth mixture ratio R5(k) and the sixth mixture ratio R6(k) based on the time change of the coefficient vector w2(k) of the adaptive filter 803 to generate a fourth mixture ratio R4(k). As a result, the signal processing apparatus according to the third embodiment can achieve both fast convergence and a low-distortion output signal without manually controlling the step sizes μ1(k) and μ2(k), even if the gain of the acoustic impulse response approximated by the adaptive filter is less than 1.

[0099] The estimation unit 806 also includes a correction unit 310, a mixer 506, a signal ratio estimation unit 503, a correction unit 710, a mixer 1106 (corresponding to an example of a fourth mixer), and a signal ratio estimation unit 1103 (corresponding to an example of a sixth signal ratio estimation unit). The correction unit 310 receives the noise estimate e2(k) and the noisy signal xP(k) as input and calculates a corrected noise estimate e2C(k). The mixer 506 mixes the corrected noise estimate e2C(k) and the pseudo-noise n1(k) based on the time variation of the coefficient vector w1(k) of the adaptive filter 203 to generate a first mixed signal n3(k). The signal ratio estimation unit 503 uses the first mixed signal n3(k) and the speech estimate e1(k) to estimate the amplitude or power ratio of speech to noise at the input terminal 201 as a first mixture ratio R1(k). The corrector 710 receives the speech estimate e1(k) and the reference signal xR(k) as input and calculates the corrected speech estimate e1C(k). The mixer 1106 mixes the corrected speech estimate e1C(k) and the pseudo crosstalk n2(k) based on the time variation of the coefficient vector w2(k) of the adaptive filter 803 to generate a second mixed signal n4(k). The signal ratio estimator 1103 uses the second mixed signal n4(k) and the noise estimate e2(k) to estimate the ratio of the amplitude or power of the noise to the crosstalk at the input terminal 202 as a fourth mixing ratio R4(k). As a result, the signal processing apparatus according to the third embodiment can achieve both fast convergence and low-distortion output signals without manually controlling the step sizes μ1(k) and μ2(k), even if the gain of the acoustic impulse response approximated by the adaptive filter is less than 1.

[0100] Furthermore, the mixer 506 sets the content ratio of the corrected noise estimate value e2C(k) in the first mixed signal n3(k) to 100% when the coefficient update of the adaptive filter 203 starts, and sets the content ratio of the corrected noise estimate value e2C(k) in the first mixed signal n3(k) to 0% when the time change of the coefficient vector w1(k) of the adaptive filter 203 becomes sufficiently small. Furthermore, the mixer 1106 sets the content ratio of the corrected speech estimate value e1C(k) in the second mixed signal n4(k) to 100% when the coefficient update of the adaptive filter 803 starts, and sets the content ratio of the corrected speech estimate value e1C(k) in the second mixed signal n4(k) to 0% when the time change of the coefficient vector w2(k) of the adaptive filter 803 becomes sufficiently small. As a result, the signal processing device of the third embodiment can achieve both fast convergence and low-distortion output signals without manually controlling the step sizes μ1(k) and μ2(k), even if the gain of the acoustic impulse response approximated by the adaptive filter is less than 1.

[0101] Furthermore, the time change in coefficient vector w1(k) of adaptive filter 203 is the time change in the sum of squares or the sum of absolute values ​​of coefficient vector w1(k), and the time change in coefficient vector w2(k) of adaptive filter 803 is the time change in the sum of squares or the sum of absolute values ​​of coefficient vector w2(k). As a result, the signal processing devices according to the first to third embodiments can achieve both fast convergence and low-distortion output signals without manually controlling the step sizes μ1(k) and μ2(k), even if the gain of the acoustic impulse response approximated by the adaptive filter is less than 1.

[0102] Furthermore, the time change of coefficient vector w1(k) of adaptive filter 203 is the time change of the partial sum of squares or the partial sum of absolute values ​​of coefficient vector w1(k), and the time change of coefficient vector w2(k) of adaptive filter 803 is the time change of the partial sum of squares or the partial sum of absolute values ​​of coefficient vector w2(k). As a result, the signal processing devices according to the first to third embodiments can achieve both fast convergence and low-distortion output signals without manually controlling the step sizes μ1(k) and μ2(k), even if the gain of the acoustic impulse response approximated by the adaptive filter is less than 1.

[0103] The above describes the embodiments of the present application in detail based on the drawings, but this is merely an example, and the present invention can be implemented in other forms that include the embodiments described in the Disclosure of the Invention section and that have been modified and improved in various ways based on the knowledge of those skilled in the art.

[0104] Furthermore, the above-mentioned "section, module, unit" can be read as "means" or "circuit," etc. For example, a subtraction section can be read as subtraction means or a subtraction circuit. (Appendix 1) a first input means for inputting a first mixed signal in which the first signal and the second signal are mixed; a second input means for inputting a second mixed signal in which a third signal correlated with the first signal and a fourth signal correlated with the second signal are mixed; a first adaptive filter (adaptive filter 103, 203) for filtering the second mixture signal to generate an estimate of the second signal; a first subtraction unit (subtraction unit 104, 204) that generates an estimate of the first signal from the first mixed signal and an estimate of the second signal; an estimation unit (estimation unit 106, 206, 806) that estimates a ratio of amplitudes or powers of the first signal and the second signal as a first mixture ratio using the estimated value of the first signal, the estimated value of the second signal, the second mixture signal, the first mixture signal, and a coefficient of the first adaptive filter; Equipped with The first adaptive filter is controlled using the first mixing ratio. Signal processing device. (Appendix 2) The estimation unit (estimation unit 106, 206) a first signal ratio estimator (signal ratio estimator 301) that estimates a ratio of amplitudes or powers of the first signal and the second signal as a second mixture ratio using the estimated value of the first signal and the estimated value of the second signal; a first correction unit (correction unit 310) that corrects the second mixture signal using the second mixture signal and the first mixture signal to generate a corrected second mixture signal; a second signal ratio estimator (signal ratio estimator 302) that estimates a ratio of amplitude or power of the first signal to the second signal as a third mixture ratio using the estimated value of the first signal and the corrected second mixture signal; a first mixer (mixer 305) that mixes the second mixture ratio and the third mixture ratio based on a time change in a coefficient of the first adaptive filter to generate the first mixture ratio; 2. The signal processing device according to claim 1, comprising: (Appendix 3) The first mixing section (mixing section 305) When coefficient update of the first adaptive filter starts, the content ratio of the third mixture ratio in the first mixture ratio is set to 100%, and when a time change in the coefficient of the first adaptive filter becomes sufficiently small, the content ratio of the third mixture ratio in the first mixture ratio is set to 0%. 3. The signal processing device of claim 2. (Appendix 4) The estimation unit (estimation unit 106, 206) a first correction unit (correction unit 310) that corrects the second mixture signal using the second mixture signal and the first mixture signal to generate a corrected second mixture signal; a second mixer (mixer 506) that mixes the corrected second mixed signal and the estimated value of the second signal based on time changes in the coefficients of the first adaptive filter to generate a first mixed signal; a third signal ratio estimator (signal ratio estimator 503) that estimates a ratio of amplitudes or powers of the first signal and the second signal as the first mixture ratio using the first mixed signal and an estimated value of the first signal; 2. The signal processing device according to claim 1, comprising: (Appendix 5) The second mixing section (mixing section 506) When the coefficient update of the first adaptive filter starts, the content ratio of the corrected second mixture signal in the first mixture signal is set to 100%, and when the change over time in the coefficient of the first adaptive filter becomes sufficiently small, the content ratio of the corrected second mixture signal in the first mixture signal is set to 0%. 5. The signal processing device of claim 4. (Appendix 6) a second adaptive filter (adaptive filter 803) that filters the estimate of the first signal to generate an estimate of the third signal; a second subtraction unit (subtraction unit 804) that subtracts the estimated value of the third signal from the second mixed signal to generate the estimated value of the fourth signal; The first adaptive filter is an estimate of the fourth signal is input instead of the second mixed signal; The estimation unit further estimating, as a fourth mixture ratio, a ratio of amplitudes or powers of the fourth signal and the third signal using the estimated value of the fourth signal, the estimated value of the third signal, the estimated value of the first signal, the second mixture signal, and a coefficient of the second adaptive filter; The second adaptive filter is controlled using the fourth mixing ratio. 2. The signal processing device of claim 1. (Appendix 7) The estimation unit (estimation unit 806) a first signal ratio estimator (signal ratio estimator 301) that estimates a ratio of amplitudes or powers of the first signal and the second signal as a second mixture ratio using the estimated value of the first signal and the estimated value of the second signal; a first correction unit (correction unit 310) that corrects the estimate of the fourth signal using the estimate of the fourth signal and the first mixed signal to generate a corrected estimate of the fourth signal; a second signal ratio estimator (signal ratio estimator 302) that estimates a ratio of amplitudes or powers of the first signal and the second signal as a third mixture ratio using the estimated value of the first signal and the corrected estimated value of the fourth signal; a first mixer (mixer 305) that mixes the second mixture ratio and the third mixture ratio based on a time change in a coefficient of the first adaptive filter to generate the first mixture ratio; a fourth signal ratio estimator (signal ratio estimator 901) that estimates a ratio of amplitudes or powers of the fourth signal and the third signal as a fifth mixture ratio using the estimated value of the fourth signal and the estimated value of the third signal; a second correction unit (correction unit 710) that corrects the estimate of the first signal using the estimate of the first signal and the second mixed signal to generate a corrected estimate of the first signal; a fifth signal ratio estimator (signal ratio estimator 902) that estimates a ratio of amplitudes or powers of the fourth signal and the third signal as a sixth mixture ratio using the estimated value of the fourth signal and the corrected estimated value of the first signal; a third mixer (mixer 905) that mixes the fifth mixture ratio and the sixth mixture ratio based on a time change in the coefficient of the second adaptive filter to generate the fourth mixture ratio; 7. The signal processing device according to claim 6, comprising: (Appendix 8) The first mixing section (mixing section 305) setting a content ratio of the third mixture ratio in the first mixture ratio to 100% when a coefficient update of the first adaptive filter starts, and setting a content ratio of the third mixture ratio in the first mixture ratio to 0% when a time change in the coefficient of the first adaptive filter becomes sufficiently small; The third mixing section (mixing section 905) The proportion of the sixth mixture ratio in the fourth mixture ratio is set to 100% when the coefficient update of the second adaptive filter starts, and the proportion of the sixth mixture ratio in the fourth mixture ratio is set to 0% when the change over time in the coefficient of the second adaptive filter becomes sufficiently small. 9. The signal processing device according to claim 7 or 8. (Appendix 9) The estimation unit (estimation unit 806) a first correction unit (correction unit 310) that corrects the estimate of the fourth signal using the estimate of the fourth signal and the first mixed signal to generate a corrected estimate of the fourth signal; a second mixer (mixer 506) that mixes the corrected estimate of the fourth signal and the estimate of the second signal based on time changes in the coefficients of the first adaptive filter to generate a first mixed signal; a third signal ratio estimator (signal ratio estimator 503) that estimates a ratio of amplitudes or powers of the first signal and the second signal as the first mixture ratio using estimated values ​​of the first mixed signal and the first signal; a second correction unit (correction unit 710) that corrects the estimate of the first signal using the estimate of the first signal and the second mixed signal to generate a corrected estimate of the first signal; a fourth mixer (mixer 1106) that mixes the corrected estimate of the first signal and the estimate of the third signal based on time variations in the coefficients of the second adaptive filter to generate a second mixed signal; a sixth signal ratio estimator (signal ratio estimator 1103) that estimates a ratio of amplitudes or powers of the fourth signal and the third signal as the fourth mixture ratio using estimated values ​​of the second mixed signal and the fourth signal; 7. The signal processing device according to claim 6, comprising: (Appendix 10) The second mixing section (mixing section 506) setting a content ratio of the corrected estimate value of the fourth signal in the first mixed signal to 100% when coefficient update of the first adaptive filter starts, and setting a content ratio of the corrected estimate value of the fourth signal in the first mixed signal to 0% when a time change in the coefficient of the first adaptive filter becomes sufficiently small; The fourth mixing unit (mixing unit 1106) When the coefficient update of the second adaptive filter starts, the content ratio of the corrected estimate value of the first signal in the second mixed signal is set to 100%, and when the time change of the coefficient of the second adaptive filter becomes sufficiently small, the content ratio of the corrected estimate value of the first signal in the second mixed signal is set to 0%. 10. The signal processing device of claim 9. (Appendix 11) The time change of the coefficient is The time change of the sum of squares or absolute values ​​of the coefficients is 11. The signal processing device of claim 3, 5, 8, or 10. (Appendix 12) The time change of the coefficient is The time change of the squared partial sum or absolute partial sum of the coefficients is 11. The signal processing device of claim 3, 5, 8, or 10. (Appendix 13) Input the first mixed signal, which is a mixture of the first and second signals, a second mixed signal in which a third signal correlated with the first signal and a fourth signal correlated with the second signal are mixed is input; processing the second mixed signal with a first adaptive filter (adaptive filter 103, 203) to generate an estimate of the second signal; generating an estimate of the first signal from the first mixed signal and an estimate of the second signal; estimating an amplitude or power ratio between the first signal and the second signal as a first mixture ratio using the estimated value of the first signal, the estimated value of the second signal, the second mixture signal, the first mixture signal, and a coefficient of the first adaptive filter; and controlling generation of an estimate of the second signal using the first mixing ratio. Signal processing methods. (Appendix 14) processing the estimate of the first signal with a second adaptive filter (adaptive filter 803) to generate an estimate of the third signal; subtracting the estimate of the third signal from the second mixed signal to generate an estimate of the fourth signal; the first adaptive filter processes an estimate of the fourth signal instead of the second mixed signal; further using the estimated value of the fourth signal, the estimated value of the third signal, the estimated value of the first signal, the second mixed signal, and a coefficient of the second adaptive filter, further estimating a ratio of amplitude or power of the fourth signal to the third signal as a fourth mixture ratio; and controlling generation of an estimate of the third signal using the fourth mixing ratio. 14. A signal processing method according to claim 13. (Appendix 15) On the computer, inputting a first mixed signal in which a first signal and a second signal are mixed; inputting a second mixed signal in which a third signal correlated with the first signal and a fourth signal correlated with the second signal are mixed; processing the second mixed signal with a first adaptive filter (adaptive filter 103, 203) to generate an estimate of the second signal; generating an estimate of the first signal from the first mixed signal and an estimate of the second signal; a step of estimating a ratio of amplitudes or powers of the first signal and the second signal as a first mixture ratio using the estimated value of the first signal, the estimated value of the second signal, the second mixture signal, the first mixture signal, and a coefficient of the first adaptive filter; using the first mixing ratio to control generation of an estimate of the second signal; A signal processing program that executes the above. (Appendix 16) On the computer, processing the estimate of the first signal with a second adaptive filter (adaptive filter 803) to generate an estimate of the third signal; subtracting the estimate of the third signal from the second mixture signal to generate an estimate of the fourth signal; the first adaptive filter processes an estimate of the fourth signal instead of the second mixed signal; further using the estimated value of the fourth signal, the estimated value of the third signal, the estimated value of the first signal, the second mixed signal, and a coefficient of the second adaptive filter, further estimating a ratio of amplitudes or powers of the fourth signal and the third signal as a fourth mixing ratio; using the fourth mixing ratio to control generation of an estimate of the third signal; 17. The signal processing program according to claim 16, [Explanation of symbols]

[0105] 100 Signal processing device 101 First input section 102 Second input section 103,203 Adaptive filter (equivalent to an example of the first adaptive filter) 104, 204 subtraction unit (corresponding to an example of a first subtraction unit) 106,206,806 Estimation part 107 Coefficient update control section 141 coefficient 200,800 Noise canceller (equivalent to an example of a signal processing device) 201 input terminal (corresponding to an example of the first input unit) 202 input terminal (corresponding to an example of the second input unit) 205,805 output terminal 301 signal ratio estimation unit (corresponding to an example of a first signal ratio estimation unit) 302 signal ratio estimation unit (corresponding to an example of a second signal ratio estimation unit) 305 Mixing section (corresponding to an example of the first mixing section) 310 correction unit (corresponding to an example of the first correction unit) 506 Mixing section (equivalent to an example of the second mixing section) 503 signal ratio estimation unit (corresponding to an example of a third signal ratio estimation unit) 710 Correction unit (corresponding to an example of the second correction unit) 803 Adaptive filter (equivalent to an example of a second adaptive filter) 804 subtraction unit (corresponding to an example of the second subtraction unit) 901 signal ratio estimation unit (corresponding to an example of a fourth signal ratio estimation unit) 902 signal ratio estimation unit (corresponding to an example of a fifth signal ratio estimation unit) 905 Mixing section (equivalent to an example of the third mixing section) 1106 Mixing section (equivalent to an example of the 4th mixing section) 1103 signal ratio estimation unit (corresponding to an example of a sixth signal ratio estimation unit) A signal source B signal source xP(k) 1st mixed signal xR(k) 2nd mixed signal xRC(k) Correction 2nd mixed signal e1(k) Estimated value of the first signal, estimated value of the speech signal e2(k) Noise estimate (equivalent to an example of the fourth signal estimate) e1C(k) Corrected estimate of the first signal, corrected estimate of the speech signal e2C(k) Corrected noise estimate (corresponding to an example of the corrected fourth signal estimate) n1(k) is an estimate of the second signal, pseudo-noise (equivalent to an example of an estimate of the second signal) n2(k) pseudo crosstalk (corresponding to an example of an estimate of the third signal) n3(k) mixed signal (corresponding to an example of the first mixed signal) n4(k) mixed signal (corresponding to an example of the second mixed signal) R1(k) 1st mixed ratio R2(k) 2nd mixed ratio R3(k) 3rd mixed ratio R4(k) 4th Mixture Ratio R5(k) 5th mixed ratio R6(k) 6th Mixture Ratio

Claims

1. a first input means for inputting a first mixed signal in which the first signal and the second signal are mixed; a second input means for inputting a second mixed signal in which a third signal correlated with the first signal and a fourth signal correlated with the second signal are mixed; a first adaptive filter for filtering the second mixed signal to generate an estimate of the second signal; a first subtraction unit that generates an estimate of the first signal from the first mixed signal and an estimate of the second signal; an estimation unit that estimates a ratio of amplitudes or powers of the first signal and the second signal as a first mixture ratio using the estimated value of the first signal, the estimated value of the second signal, the second mixture signal, the first mixture signal, and a coefficient of the first adaptive filter; Equipped with controlling the first adaptive filter using the first mixture ratio; The estimation unit a first signal ratio estimator that estimates a ratio of amplitudes or powers of the first signal and the second signal as a second mixture ratio using the estimated value of the first signal and the estimated value of the second signal; a first correction unit that averages the power of the second mixture signal and the first mixture signal, sets a ratio between the averaged second mixture signal and the first mixture signal as a magnification, and generates a corrected second mixture signal by multiplying the square root of the magnification by the second mixture signal; a second signal ratio estimator that estimates a ratio of amplitudes or powers of the first signal and the second signal as a third mixture ratio using the estimated value of the first signal and the corrected second mixture signal; a first mixer that mixes the second mixture ratio and the third mixture ratio by setting a weight of the third mixture ratio to a large value at the start of updating the coefficients and decreasing the weight as the coefficients grow, to generate the first mixture ratio; A signal processing device comprising:

2. The first mixing section When coefficient update of the first adaptive filter is started, the content ratio of the third mixture ratio in the first mixture ratio is set to 100%, and when a time change in the coefficient of the first adaptive filter becomes sufficiently small, the content ratio of the third mixture ratio in the first mixture ratio is set to 0%. The signal processing device according to claim 1 .

3. A first input means for inputting a first mixed signal in which a first signal and a second signal are mixed; a second input means for inputting a second mixed signal in which a third signal correlated with the first signal and a fourth signal correlated with the second signal are mixed; a first adaptive filter for filtering the second mixed signal to generate an estimate of the second signal; a first subtraction unit that generates an estimate of the first signal from the first mixed signal and an estimate of the second signal; an estimation unit that estimates a ratio of amplitudes or powers of the first signal and the second signal as a first mixture ratio using the estimated value of the first signal, the estimated value of the second signal, the second mixture signal, the first mixture signal, and a coefficient of the first adaptive filter; Equipped with controlling the first adaptive filter using the first mixture ratio; The estimation unit a first correction unit that averages the power of the second mixture signal and the first mixture signal, sets a ratio between the averaged second mixture signal and the first mixture signal as a magnification, and generates a corrected second mixture signal by multiplying the square root of the magnification by the second mixture signal; a second mixer that mixes the corrected second mixture signal and the estimated value of the second signal by setting a weight of the corrected second mixture signal to a large value when the coefficient update starts and decreasing the weight as the coefficient grows, to generate a first mixture signal; a third signal ratio estimator that estimates, as the first mixture ratio, a ratio of amplitudes or powers of the first signal and the second signal using the first mixed signal and an estimated value of the first signal; A signal processing device comprising:

4. The second mixing section is When the coefficient update of the first adaptive filter starts, the content ratio of the corrected second mixture signal in the first mixture signal is set to 100%, and when the change over time in the coefficient of the first adaptive filter becomes sufficiently small, the content ratio of the corrected second mixture signal in the first mixture signal is set to 0%. The signal processing device according to claim 3 .

5. a second adaptive filter that filters the estimate of the first signal to generate an estimate of the third signal; a second subtraction unit that subtracts the estimated value of the third signal from the second mixed signal to generate the estimated value of the fourth signal, The first adaptive filter is an estimate of the fourth signal is input instead of the second mixed signal; The estimation unit further estimating, as a fourth mixture ratio, a ratio of amplitudes or powers of the fourth signal and the third signal using the estimated value of the fourth signal, the estimated value of the third signal, the estimated value of the first signal, the second mixture signal, and a coefficient of the second adaptive filter; The second adaptive filter is controlled using the fourth mixing ratio.

4. The signal processing device according to claim 1 or 3.

6. The estimation unit a first signal ratio estimator that estimates a ratio of amplitudes or powers of the first signal and the second signal as a second mixture ratio using the estimated value of the first signal and the estimated value of the second signal; a first correction unit that corrects the estimate of the fourth signal using the estimate of the fourth signal and the first mixed signal to generate a corrected estimate of the fourth signal; a second signal ratio estimator that estimates a ratio of amplitudes or powers of the first signal and the second signal as a third mixture ratio using the estimated value of the first signal and the corrected estimated value of the fourth signal; a first mixer that mixes the second mixture ratio and the third mixture ratio based on a time change in a coefficient of the first adaptive filter to generate the first mixture ratio; a fourth signal ratio estimator that estimates a ratio of amplitudes or powers of the fourth signal and the third signal as a fifth mixture ratio using the estimated value of the fourth signal and the estimated value of the third signal; a second correction unit that corrects the estimate of the first signal using the estimate of the first signal and the second mixed signal to generate a corrected estimate of the first signal; a fifth signal ratio estimator that estimates a ratio of amplitudes or powers of the fourth signal and the third signal as a sixth mixture ratio using the estimated value of the fourth signal and the corrected estimated value of the first signal; a third mixer that mixes the fifth mixture ratio and the sixth mixture ratio based on a time change in a coefficient of the second adaptive filter to generate the fourth mixture ratio; The signal processing device according to claim 5 , comprising:

7. The first mixing section setting a content ratio of the third mixture ratio in the first mixture ratio to 100% when a coefficient update of the first adaptive filter starts, and setting a content ratio of the third mixture ratio in the first mixture ratio to 0% when a time change in the coefficient of the first adaptive filter becomes sufficiently small; The third mixing section is When coefficient update of the second adaptive filter is started, the content ratio of the sixth mixture ratio in the fourth mixture ratio is set to 100%, and when a time change in the coefficient of the second adaptive filter becomes sufficiently small, the content ratio of the sixth mixture ratio in the fourth mixture ratio is set to 0%. The signal processing device according to claim 6 .

8. The estimation unit a first correction unit that corrects the estimate of the fourth signal using the estimate of the fourth signal and the first mixed signal to generate a corrected estimate of the fourth signal; a second mixer that mixes the corrected estimate of the fourth signal and the estimate of the second signal based on time changes in coefficients of the first adaptive filter to generate a first mixed signal; a third signal ratio estimator that estimates a ratio of amplitudes or powers of the first signal and the second signal as the first mixture ratio using estimated values ​​of the first mixed signal and the first signal; a second correction unit that corrects the estimate of the first signal using the estimate of the first signal and the second mixed signal to generate a corrected estimate of the first signal; a fourth mixer that mixes the corrected estimate of the first signal and the estimate of the third signal based on time changes in coefficients of the second adaptive filter to generate a second mixed signal; a sixth signal ratio estimator that estimates a ratio of amplitudes or powers of the fourth signal and the third signal as the fourth mixture ratio using estimated values ​​of the second mixed signal and the fourth signal; The signal processing device according to claim 5 , comprising:

9. The second mixing section is setting a content ratio of the corrected estimate value of the fourth signal in the first mixed signal to 100% when coefficient update of the first adaptive filter starts, and setting a content ratio of the corrected estimate value of the fourth signal in the first mixed signal to 0% when a time change in the coefficient of the first adaptive filter becomes sufficiently small; The fourth mixing section is When coefficient updating of the second adaptive filter starts, the content ratio of the corrected estimated value of the first signal in the second mixed signal is set to 100%, and when time change in the coefficient of the second adaptive filter becomes sufficiently small, the content ratio of the corrected estimated value of the first signal in the second mixed signal is set to 0%. The signal processing device according to claim 8 .

10. The time change of the coefficient is The time change of the sum of squares or the sum of absolute values ​​of the coefficients is 10. A signal processing device according to claim 2, 4, 7 or 9.

11. The time change of the coefficient is is the time change of the squared partial sum or absolute partial sum of the coefficients 10. A signal processing device according to claim 2, 4, 7 or 9.

12. A first mixed signal in which the first signal and the second signal are mixed is input; a second mixed signal in which a third signal correlated with the first signal and a fourth signal correlated with the second signal are mixed is input; processing the second mixed signal with a first adaptive filter to generate an estimate of the second signal; generating an estimate of the first signal from the first mixed signal and an estimate of the second signal; estimating a ratio of amplitudes or powers of the first signal and the second signal as a second mixture ratio using the estimated value of the first signal and the estimated value of the second signal; averaging the power of the second mixture signal and the first mixture signal, determining a ratio between the averaged second mixture signal and the first mixture signal as a magnification, and generating a corrected second mixture signal by multiplying the square root of the magnification by the second mixture signal; using the estimated value of the first signal and the corrected second mixture signal, estimating an amplitude or power ratio between the first signal and the second signal as a third mixture ratio; generating a first mixture ratio by mixing the second mixture ratio and the third mixture ratio by setting a weight of the third mixture ratio to a large value at the start of updating the coefficients of the first adaptive filter and decreasing the weight as the coefficients grow; and controlling generation of an estimate of the second signal using the first mixing ratio. Signal processing methods.

13. A first mixed signal in which a first signal and a second signal are mixed is input, a second mixed signal in which a third signal correlated with the first signal and a fourth signal correlated with the second signal are mixed is input; processing the second mixed signal with a first adaptive filter to generate an estimate of the second signal; generating an estimate of the first signal from the first mixed signal and an estimate of the second signal; averaging the power of the second mixture signal and the first mixture signal, determining a ratio between the averaged second mixture signal and the first mixture signal as a magnification, and generating a corrected second mixture signal by multiplying the square root of the magnification by the second mixture signal; generating a first mixed signal by mixing the corrected second mixed signal and the estimated value of the second signal, by setting a weight of the corrected second mixed signal to a large value when updating the coefficients of the first adaptive filter starts and decreasing the weight as the coefficients grow; using the first mixed signal and the estimated value of the first signal, estimating an amplitude or power ratio between the first signal and the second signal as a first mixture ratio; and controlling generation of an estimate of the second signal using the first mixing ratio. Signal processing methods.

14. processing the estimate of the first signal with a second adaptive filter to generate an estimate of the third signal; subtracting the estimate of the third signal from the second mixed signal to generate an estimate of the fourth signal; the first adaptive filter processes an estimate of the fourth signal instead of the second mixed signal; further using the estimated value of the fourth signal, the estimated value of the third signal, the estimated value of the first signal, the second mixed signal, and a coefficient of the second adaptive filter, further estimating a ratio of amplitude or power of the fourth signal to the third signal as a fourth mixture ratio; and controlling generation of an estimate of the third signal using the fourth mixing ratio.

14. A signal processing method according to claim 12 or 13.

15. On the computer, inputting a first mixed signal in which the first signal and the second signal are mixed; inputting a second mixed signal in which a third signal correlated with the first signal and a fourth signal correlated with the second signal are mixed; processing the second mixed signal with a first adaptive filter to generate an estimate of the second signal; generating an estimate of the first signal from the first mixed signal and an estimate of the second signal; estimating a ratio of amplitudes or powers of the first signal and the second signal as a second mixture ratio using the estimated value of the first signal and the estimated value of the second signal; a step of averaging the power of the second mixture signal and the first mixture signal, determining a ratio of the averaged second mixture signal to the first mixture signal as a scaling factor, and generating a corrected second mixture signal by multiplying the square root of the scaling factor by the second mixture signal; estimating a ratio of amplitudes or powers of the first signal and the second signal as a third mixture ratio using the estimated value of the first signal and the corrected second mixture signal; generating a first mixture ratio by mixing the second mixture ratio and the third mixture ratio by setting a weight of the third mixture ratio to a large value at the start of updating the coefficients of the first adaptive filter and decreasing the weight as the coefficients grow; using the first mixing ratio to control generation of an estimate of the second signal; A signal processing program that executes the above.

16. A computer comprising: inputting a first mixed signal in which the first signal and the second signal are mixed; inputting a second mixed signal in which a third signal correlated with the first signal and a fourth signal correlated with the second signal are mixed; processing the second mixed signal with a first adaptive filter to generate an estimate of the second signal; generating an estimate of the first signal from the first mixed signal and an estimate of the second signal; a step of averaging the power of the second mixture signal and the first mixture signal, determining a ratio of the averaged second mixture signal to the first mixture signal as a scaling factor, and generating a corrected second mixture signal by multiplying the square root of the scaling factor by the second mixture signal; generating a first mixed signal by mixing the corrected second mixed signal and the estimated value of the second signal, by setting a weight of the corrected second mixed signal to a large value when updating the coefficients of the first adaptive filter starts and decreasing the weight as the coefficients grow; a step of estimating a ratio of amplitudes or powers of the first signal and the second signal as a first mixture ratio using the first mixed signal and the estimated value of the first signal; using the first mixing ratio to control generation of an estimate of the second signal; A signal processing program that executes the above.

17. On the computer, processing the estimate of the first signal with a second adaptive filter to generate an estimate of the third signal; subtracting the estimate of the third signal from the second mixture signal to generate an estimate of the fourth signal; the first adaptive filter processes an estimate of the fourth signal instead of the second mixed signal; further using the estimated value of the fourth signal, the estimated value of the third signal, the estimated value of the first signal, the second mixed signal, and a coefficient of the second adaptive filter, further estimating a ratio of amplitude or power of the fourth signal to the third signal as a fourth mixing ratio; using the fourth mixing ratio to control generation of an estimate of the third signal; 17. The signal processing program according to claim 15, wherein the signal processing program executes the following:

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