WO2012153451A1 - 適応等化器、音響エコーキャンセラ装置および能動騒音制御装置 - Google Patents
適応等化器、音響エコーキャンセラ装置および能動騒音制御装置 Download PDFInfo
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
- G10K11/00—Methods or devices for transmitting, conducting or directing sound in general; Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
- G10K11/16—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
- G10K11/175—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
- G10K11/00—Methods or devices for transmitting, conducting or directing sound in general; Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
- G10K11/16—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
- G10K11/175—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
- G10K11/178—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound by electro-acoustically regenerating the original acoustic waves in anti-phase
- G10K11/1785—Methods, e.g. algorithms; Devices
- G10K11/17853—Methods, e.g. algorithms; Devices of the filter
- G10K11/17854—Methods, e.g. algorithms; Devices of the filter the filter being an adaptive filter
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
- G10K11/00—Methods or devices for transmitting, conducting or directing sound in general; Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
- G10K11/16—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
- G10K11/175—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
- G10K11/178—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound by electro-acoustically regenerating the original acoustic waves in anti-phase
- G10K11/1785—Methods, e.g. algorithms; Devices
- G10K11/17855—Methods, e.g. algorithms; Devices for improving speed or power requirements
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
- G10K11/00—Methods or devices for transmitting, conducting or directing sound in general; Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
- G10K11/16—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
- G10K11/175—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
- G10K11/178—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound by electro-acoustically regenerating the original acoustic waves in anti-phase
- G10K11/1787—General system configurations
- G10K11/17879—General system configurations using both a reference signal and an error signal
- G10K11/17881—General system configurations using both a reference signal and an error signal the reference signal being an acoustic signal, e.g. recorded with a microphone
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- H—ELECTRICITY
- H03—ELECTRONIC CIRCUITRY
- H03G—CONTROL OF AMPLIFICATION
- H03G5/00—Tone control or bandwidth control in amplifiers
- H03G5/16—Automatic control
- H03G5/165—Equalizers; Volume or gain control in limited frequency bands
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04B—TRANSMISSION
- H04B3/00—Line transmission systems
- H04B3/02—Details
- H04B3/20—Reducing echo effects or singing; Opening or closing transmitting path; Conditioning for transmission in one direction or the other
- H04B3/23—Reducing echo effects or singing; Opening or closing transmitting path; Conditioning for transmission in one direction or the other using a replica of transmitted signal in the time domain, e.g. echo cancellers
- H04B3/235—Reducing echo effects or singing; Opening or closing transmitting path; Conditioning for transmission in one direction or the other using a replica of transmitted signal in the time domain, e.g. echo cancellers combined with adaptive equaliser
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R3/00—Circuits for transducers, loudspeakers or microphones
- H04R3/02—Circuits for transducers, loudspeakers or microphones for preventing acoustic reaction, i.e. acoustic oscillatory feedback
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
- G10K2210/00—Details of active noise control [ANC] covered by G10K11/178 but not provided for in any of its subgroups
- G10K2210/30—Means
- G10K2210/301—Computational
- G10K2210/3014—Adaptive noise equalizers [ANE], i.e. where part of the unwanted sound is retained
Definitions
- the present invention relates to an adaptive equalizer used for learning identification of an unknown system, an acoustic echo canceller device using the adaptive equalizer, and an active noise control device.
- An adaptive equalizer or adaptive filter has a function to identify the transfer function of an unknown system from an input signal to the target unknown system and its response signal using an adaptive algorithm, and is widely used in various signal processing systems. It is used for.
- an echo signal is predicted from a signal that is the source of an echo by identifying a transfer function of the echo path, and is used to cancel the echo signal.
- the active noise control device predicts the incoming noise from the acoustic signal of the noise source by identifying the transfer function of the noise path, and cancels the noise by generating an acoustic signal that is opposite in phase to this. Used.
- the adaptive equalizer when canceling the noise of the blower fan in the blower duct, the sound other than the blower fan picked up by the error detection microphone provided in the duct (operation sound of another mechanical device, voice of a nearby person, etc.) For the adaptive equalizer, it becomes a disturbance in identifying the noise path, and becomes a factor of reducing the noise reduction effect.
- Equation (1) N in equation (2) is the filter order.
- n a subscript representing a time series.
- ⁇ represents the update step size
- ⁇ (n) represents the update amount of the filter coefficient given by a predetermined adaptive algorithm.
- ⁇ (n) when a generally well-known NLMS (learning identification method) algorithm is used, it is expressed by the following equation (3).
- x (n) is an input signal and is expressed by the following equation (4).
- the right side of Expression (5) is the signal power of the input signal x (n), and d (n) is a residual signal obtained by the filter coefficient before update.
- equation (1) by setting the update step size ⁇ to a small value, the coefficient update speed can be delayed, and the influence of disturbance can be made difficult. As a result, the convergence value of the identification error (filter coefficient error) can be made smaller, but on the other hand, since the coefficient update becomes slow, the transfer function of the unknown system changes when starting from the initial state. Immediately after, for example, more update times are required until the identification error converges. For this reason, the update step size ⁇ needs to be set so as to obtain a convergence characteristic according to the purpose.
- the update step size ⁇ is often a constant, but can be changed according to the situation.
- Patent Document 1 proposes an echo canceller that changes the update step size ⁇ so that it always converges to a required identification error regardless of the disturbance condition.
- the update step size ⁇ is determined according to the required value of the identification error, so that the update step size also increases or decreases depending on the required value. As a result, as the smaller identification error is desired, the value of the update step size ⁇ becomes smaller, which slows down the coefficient update. Problem arises.
- Non-Patent Document 1 an accurate ⁇ opt cannot be obtained due to an error in the observed cross-correlation value, and an update step size that is slightly higher or lower than this is calculated. As a result, it becomes impossible to reduce the time to a sufficiently low level.
- the update step size value is determined according to the required value of the identification error. Therefore, the smaller the required identification error, the more the number of filter coefficient updates required to converge the identification error. There is a problem that it takes time to converge.
- the optimal update step size value is not set due to the cross-correlation value observation error, and the identification error is quickly There was a problem that it could not be reduced to a sufficient level. Such a problem causes a problem that, for example, the transfer function of the echo path cannot be identified quickly or sufficiently accurately in an echo canceller, and a sufficient echo cancellation effect cannot be obtained.
- the active noise control device there is a problem that the noise transmission path cannot be identified quickly or accurately, and a sufficient noise suppression effect cannot be obtained.
- the present invention has been made to solve the above-mentioned problems, and provides means for observing or detecting disturbances, states of transfer functions of unknown systems, and conditions in which changes cannot be predicted in advance.
- the purpose is to determine a variable update step size according to these conditions without using them separately.
- the adaptive equalizer according to the present invention arbitrarily selects the first residual signal obtained using the adaptive filter coefficient sequence obtained by the previous operation and the adaptive filter coefficient sequence obtained by the previous operation.
- Variable update step determined in proportion to the size ratio or the size difference with the second residual signal obtained using the pre-update adaptive filter coefficient sequence that has been updated using the pre-update step size.
- An input signal is filtered using an adaptive filter coefficient sequence that has been subjected to a coefficient update process using the size to generate an output signal.
- the identification error can be quickly reduced to a sufficiently small level.
- FIG. 3 is a block diagram showing a configuration of an adaptive equalizer according to Embodiment 1.
- FIG. 10 is an explanatory diagram illustrating a read / write process of a filter coefficient sequence of the adaptive equalizer according to the first embodiment.
- FIG. 10 is a block diagram showing another configuration of the adaptive equalizer according to the first embodiment.
- 4 is a graph showing an example of identification error convergence characteristics of the adaptive equalizer according to the first embodiment.
- FIG. 10 is a block diagram showing a configuration of an adaptive equalizer according to a second embodiment.
- FIG. 10 is an explanatory diagram showing read / write processing of a filter coefficient sequence of an adaptive equalizer according to a second embodiment.
- FIG. 10 is a block diagram showing a configuration of an adaptive equalizer according to a third embodiment.
- FIG. 10 is an explanatory diagram showing read / write processing of a filter coefficient sequence of an adaptive equalizer according to a third embodiment.
- FIG. 10 is a block diagram showing a configuration of an adaptive equalizer according to a fourth embodiment.
- FIG. 10 is an explanatory diagram illustrating read / write processing of a filter coefficient sequence of an adaptive equalizer according to a fourth embodiment.
- FIG. 10 is a block diagram showing a configuration of an adaptive equalizer according to a fifth embodiment.
- FIG. 10 is a block diagram showing a configuration of an acoustic echo canceller apparatus according to a sixth embodiment. It is a block diagram which shows the structure of the active noise control apparatus by Embodiment 7.
- FIG. 7 shows the structure of the active noise control apparatus by Embodiment 7.
- FIG. 1 is a block diagram showing a configuration of an adaptive equalizer according to the first embodiment.
- the adaptive equalizer 100 updates the coefficients of the subtraction filter 101 that generates the first estimated response signal and the first residual signal and the filter coefficient sequence, and uses the updated filter coefficient sequence to generate the second
- the first update filter 102 for generating the estimated response signal
- the step size control unit 103 for determining the variable update step size
- the second update filter 104 for updating the coefficient of the filter coefficient sequence
- the memory unit for storing the filter coefficient sequence 105.
- the subtraction filter 101 includes an estimated response signal generation unit 101a and a subtractor 101b
- the first update filter 102 includes a filter processing unit 102a, a multiplier 102b, and a subtractor 102c
- the second update filter 104 updates a coefficient.
- a processing unit 104a and a multiplier 104b are provided.
- the adaptive equalizer 100 is connected to an unknown system 900.
- the unknown system 900 is a system that is a learning identification target of the adaptive equalizer 100.
- FIG. 2 is an explanatory diagram schematically showing reading or writing processing of the filter coefficient sequence of the adaptive equalizer according to the first embodiment.
- the step size control unit 103 determines the variable update step size ⁇ opt based on the first residual signal d 1 (n), the second residual signal d 2 (n), and the preceding update step size ⁇ prio (n). (N) is defined. Details of how to determine the variable update step size ⁇ opt (n) will be described later.
- variable update step size ⁇ opt (n) is defined as “update step size that minimizes the identification error in the disturbance condition and the convergence state” at each time point, and this is ensured in each coefficient update. It is characterized by having a configuration required in In order to explain the configuration, first, the derivation of the variable update step size ⁇ opt (n) will be described below.
- Equation (16) indicates that the difference in the magnitude of the identification error before and after the coefficient update is a quadratic function of the update step size ⁇ , which means that the identification error decreases if the right side is a negative value. is doing. Therefore, the update step size ⁇ that gives the minimum value on the right side of the equation (16) becomes “update step size that minimizes the identification error among disturbance conditions and convergence states” ⁇ opt , which is expressed by the following equation (17 ).
- the step size control unit 103 uses the first residual signal d 1 (n), the second residual signal d 2 (n), and the preceding update step size ⁇ prio (n). Then, the variable update step size ⁇ opt (n) represented by the following equation (22) is calculated for each time point n.
- p 1 is a function representing the magnitude of the first residual signal d 1 (n)
- P 2 is a function representing the magnitude of the second residual signal d 2 (n). Integration can be applied. Dispersion measurement using these methods can be expected to be relatively accurate compared to correlation measurement, even if the signal length is limited.
- equation (23) may be calculated in a simplified manner as the following equation (26) by setting the preceding update step size ⁇ prio (n) to 1.
- the second update filter 104 determines “the identification error in the disturbance condition and the convergence state at each time point”. It is possible to update the coefficient by reliably calculating the “update step size that reduces the most” every time. As a result, it is possible to obtain the adaptive equalizer 100 in which the identification error is rapidly reduced and the convergence value of the identification error is sufficiently small.
- the update step size ⁇ in which the identification error is reduced is a range in which the right side of equation (16) is a negative value. Therefore, the update step size ⁇ ′ opt (n) that satisfies it is satisfied. It can be expected that at least the identification error will be reduced. This range is expressed as the following equation (27).
- the adaptive equalizer of the present invention includes such a case.
- FIG. 3 is a block diagram illustrating another configuration example of the adaptive equalizer according to the first embodiment.
- the adaptive equalizer 110 in FIG. 3 is provided with a third update filter 106 in addition to the adaptive equalizer 100 shown in FIG.
- the third update filter 106 includes a filter processing unit 106a, a multiplier 106b, and a subtractor 106c, and the operation is the same as that of the first update filter 102.
- the first update filter 102 uses the first preceding update step size ⁇ prio1 (n)
- the third update filter 106 uses the second preceding update step size ⁇ prio2 (n).
- the step size control unit 103 has three residual signals, that is, the first residual signal d 1 (n) from the subtraction filter 101, the first The second residual signal d 2 (n) from the update filter 102 and the third residual signal d 3 (n) from the third update filter 106 are input.
- the step size control unit 103 determines the three residual signals of the first residual signal d 1 (n), the second residual signal d 2 (n), and the third residual signal d 3 (n). Any two of the residual signals d A (n), d B (n) and the first preceding update step size ⁇ prio1 (n), or the second preceding update step size ⁇ prio2 (n), or A variable update step size ⁇ opt (n) is determined from any two parameters ⁇ A and ⁇ B among three different parameters “0” based on the following equation (28).
- the step size control unit 103 calculates the variable update step size ⁇ opt1 (n) calculated from the pattern 1, the variable update step size ⁇ opt2 (n) calculated from the pattern 2, and the variable update step size calculated from the pattern 3.
- One of ⁇ opt3 (n) may be used as the final variable update step size ⁇ opt (n), or an average value of the three calculated variable update step sizes is calculated, and the final variable update step size ⁇ opt is calculated. (N) may be used.
- the step size control unit 103 that calculates an average value of a plurality of variable update step sizes as a variable update step size ⁇ opt (n) as a final result using a plurality of residual signals of three or more is provided.
- a highly accurate variable update step size ⁇ opt (n) can be obtained.
- a third update filter 106 is additionally provided, and the first residual signal d 1 (n), the second residual signal d 2 (n), and the third residual signal d 3 are provided.
- the configuration in which (n) is input to the step size control unit 103 is shown, the number of update filters to be added and the number of residual signals d input to the step size control unit 103 can be changed as appropriate.
- variable update step size ⁇ opt (n) when the LMS algorithm and the affine projection algorithm are used as the adaptive algorithm is as follows.
- adaptive equalizers 100 and 110 using block adaptive filter algorithms such as BLMS (Block LMS) and BOP (Block Orthogonal Projection Algorithm), which execute the LMS algorithm and the affine projection algorithm in units of predetermined signal blocks.
- block adaptive filter algorithms such as BLMS (Block LMS) and BOP (Block Orthogonal Projection Algorithm)
- the step size control unit 103 can analyze a signal having a certain block length with respect to the first residual signal d 1 (n) and the second residual signal d 2 (n). Therefore, it is possible to improve the measurement accuracy of such dispersion. This can be expected to obtain a more accurate variable update step size.
- the adaptive equalizer of the present invention includes a case where it is configured in any of the above-described examples.
- FIG. 4 is a graph showing an example of the identification error convergence characteristic by the adaptive equalizer of the first embodiment.
- FIG. 4A shows the identification error as a value obtained by normalizing the sum of the square error of the coefficients with the sum of the square values of the filter coefficients of the transfer function of the unknown system.
- the convergence characteristic curve (1) in FIG. 4 is a convergence characteristic when the adaptive equalizers 100 and 110 of the first embodiment use the variable update step size ⁇ opt (n) determined based on the above-described equation (22). Is shown.
- the convergence characteristic curve (2) has a fixed update step size of 0.05
- the convergence characteristic curve (3) has a fixed update step size of 0.1
- the convergence characteristic curve (4) has an updated step size.
- the convergence characteristics when a fixed value of 0.2 is set are shown.
- the disturbance condition is constant, and a certain level of white noise is applied to the observation response signal.
- the adaptive equalizers 100 and 110 of the first embodiment reduce the identification error more quickly than when any fixed value is used for the update step size. And the effect of converging the identification error to a small value.
- FIG. 4B shows an example of the identification error convergence characteristics when disturbance fluctuations and unknown system fluctuations occur.
- the numbers attached to the convergence characteristic curves on the graph shown in FIG. 4B are the same as those in FIG.
- constant white noise similar to that in FIG. 4A is applied as disturbance, and white noise slightly larger than other sections is further applied as sudden disturbance fluctuation in section A. is doing.
- the transfer function of the unknown system is instantaneously changed to another transfer function.
- the identification error increases due to sudden disturbance fluctuations when a fixed value of 0.05 to 0.2 is used as the update step size.
- the convergence characteristic curve (1) of the adaptive equalizers 100 and 110 maintains the convergence state without increasing the error.
- the convergence characteristic curve (1) of the adaptive equalizers 100 and 110 according to the first embodiment has the error decreasing faster than any of the other curves. It turns out that there is an effect.
- the execution or stop of coefficient update is switched according to the disturbance condition, or an unknown system change is detected and the adaptive equalizer is accelerated again. Note that no configuration is provided.
- the characteristics of the adaptive equalizers 100 and 110 according to the first embodiment shown in FIG. 4 are such that, for example, the echo canceller quickly identifies the transfer function of the echo path without being affected by disturbances such as background noise and speaker speech. This means that the echo can be effectively canceled. It also means that even when the echo path changes, the change can be quickly followed.
- the variable update step size ⁇ opt (n) that minimizes the identification error in the disturbance condition and the convergence state at each time point is calculated.
- a step size control unit 103 and a second update filter 104 that performs coefficient update using the variable update step size ⁇ opt (n) calculated by the step size control unit 103 and updates the filter coefficient sequence are configured. Therefore, the identification error can be reduced quickly and sufficiently small. Furthermore, there is an effect that the convergence state is not deteriorated even when a disturbance change that is difficult to detect occurs without separately providing a configuration for detecting a disturbance signal or a configuration for estimating the detected disturbance signal. . Furthermore, without providing a separate configuration for detecting a change in the transfer function of the unknown system 900, even if the unknown system 900 changes and the convergence state is reset, the identification error can be quickly detected by updating the filter coefficient sequence. It can be converged.
- the update step size control unit 103 includes the first residual signal d 1 (n), the second residual signal d 2 (n), and the preceding update step size ⁇ prio ( Since the variable update step size ⁇ opt (n) is calculated using n), the error of the variable update step size calculation diverges compared to the case where the preceding update step size is fixed to “1”. Thus, it is possible to suppress the failure of the learning system.
- the block adaptive filter algorithm for performing coefficient updating in units of a predetermined block length such as BLMS and BOP is applicable, the first residual signal d 1 (n) and A signal length sufficient for observing the variance of the second residual signal d 2 (n) can be obtained, and a highly accurate variable update step size ⁇ opt (n) can be calculated. Updates can be implemented.
- the quasi-variable update step size ⁇ ′ opt (n) having a predetermined range is determined, the safety of updating the filter coefficient sequence can be improved.
- the variable update step size ⁇ opt (n) is set to a small value within a predetermined range, or there is a request to increase the convergence effect.
- the variable update step size ⁇ opt (n) can be set according to the user's intention such as setting a large value within a predetermined range.
- Embodiment 2 FIG. In the first embodiment described above, after the variable update step size ⁇ opt (n) is calculated, the calculated variable update step size ⁇ opt (n) is applied to the second update filter 104, and the filter coefficient sequence A configuration for coefficient update is shown.
- the previously calculated variable update step size ⁇ opt (n ⁇ 1) and the latest variable update step size ⁇ opt (n) There are cases where there is no significant difference in value. For example, if the disturbance is white noise, pink noise, or similar random signal with a certain statistical property, the change in disturbance can be assumed to be small enough with respect to the coefficient update frequency.
- the value of the update step size is expected to hardly change between the previous time and the latest.
- the second embodiment shows a configuration in which the preceding update step size ⁇ prio (n) is set to the variable update step size ⁇ opt (n ⁇ 1) obtained in the previous operation.
- FIG. 5 is a block diagram showing a configuration of an adaptive equalizer according to the second embodiment.
- the adaptive equalizer 200 updates the coefficients of the subtraction filter 101 that generates the first estimated response signal and the first residual signal and the filter coefficient sequence, and uses the updated filter coefficient sequence to generate the second
- the memory unit 105 stores a filter coefficient sequence.
- the subtraction filter 101 includes an estimated response signal generation unit 101a and a subtractor 101b
- the first update filter 201 includes a coefficient update processing unit 201a, a multiplier 201b, and a subtractor 201c.
- the adaptive equalizer 200 is connected to the unknown system 900.
- symbol is attached
- the step size control unit 103 that obtains the variable update step size ⁇ opt (n) and the variable update step size ⁇ that is obtained by the step size control unit 103.
- opt (n) is temporarily stored and given to the first update filter 201 at the time of the next coefficient update, and the previous variable update step size ⁇ opt (n ⁇ 1) given by the delay processor 202 ) Is used to update the coefficient of the filter coefficient sequence and the first update filter 201 that generates the second estimated response signal is provided, so that the adaptive equalizer is configured with fewer components. It is possible to reduce the cost of equipment by reducing the necessary computing resources.
- Embodiment 3 the first update filter 102 immediately updates the same input signal after performing the coefficient update of the preceding update filter coefficient sequence from the input signal x (n) and the observation response signal y ′ (n).
- a configuration is shown in which a second residual signal d 2 (n) is obtained using x (n) and the observed response signal y ′ (n), and this is used to calculate the variable update step size ⁇ opt (n). It was.
- FIG. 7 is a block diagram showing a configuration of an adaptive equalizer according to the third embodiment.
- the adaptive equalizer 300 generates a first subtraction filter 301 that generates a first estimated response signal and a first residual signal, a second estimated response signal, and a second residual signal.
- a delay processing unit 306 that gives the updated coefficient amount to the update filter 307, an update filter 307 that updates the coefficient of the filter coefficient sequence, and a memory unit 308 that stores the filter coefficient sequence.
- the first subtraction filter 301 includes an estimated response signal generation unit 301a and a subtractor 301b
- the second subtraction filter 302 includes an estimation response signal generation unit 302a and a subtractor 302b
- the preceding coefficient update unit 305 calculates a coefficient.
- the update filter 307 includes a filter processing unit 307a, a multiplier 307b, and a subtractor 307c.
- the adaptive equalizer 300 is connected to the unknown system 900.
- the step size control unit 303 is based on the first residual signal d 1 (n), the second residual signal d 2 (n), and the previous update step size ⁇ prio (n ⁇ 1) used last time.
- the variable update step size ⁇ opt (n) is calculated.
- the calculation method is the same as in the first embodiment. As described in the first embodiment, when the LMS algorithm, the affine projection algorithm, or the like is used for the adaptive algorithm, the input signal x (n) is also used.
- the configuration using the coefficient update amount ⁇ (n ⁇ 1) obtained in the previous operation is shown, but this is limited to the coefficient update amount obtained in the previous operation. It is not a thing and it can change suitably.
- the coefficient update amount ⁇ (n ⁇ 2) obtained in the previous operation may be used.
- Embodiment 4 When the disturbance condition and the convergence state of the adaptive equalizer are different for each frequency band, the adaptive equalizer of the present invention is applied to the adaptive filter algorithm using time-frequency conversion such as high-speed LMS, and the disturbance condition and the convergence for each frequency band.
- time-frequency conversion such as high-speed LMS
- the adaptive equalizer of the present invention is applied to the adaptive filter algorithm using time-frequency conversion such as high-speed LMS, and the disturbance condition and the convergence for each frequency band.
- time-frequency conversion such as high-speed LMS
- FIG. 9 is a block diagram showing a configuration of an adaptive equalizer according to the fourth embodiment.
- the adaptive equalizer 400 includes a subtraction filter 401 that generates a first estimated response signal and a first residual signal, first and second time frequency conversion units 402 and 403, and coefficients of a filter coefficient sequence.
- a first update filter 404 for updating, third and fourth time frequency conversion units 405 and 407, a step size control unit 408 for determining a variable update step size, a second update filter 410 for updating a filter coefficient, and a subtractor 406, a multiplier 409, and a memory unit 411 that stores a filter coefficient sequence.
- the subtraction filter 401 includes an estimated response signal generation unit 401a and a subtractor 401b
- the first update filter 404 includes a filter processing unit 404a and a multiplier 404b
- the second update filter 410 includes a coefficient update processing unit 410a. I have.
- the adaptive equalizer 400 is connected to the unknown system 900.
- FIG. 10 is an explanatory diagram illustrating the reading or writing process of the filter coefficient sequence of the appropriate equalizer according to the fourth embodiment.
- the adaptive equalizer 400 according to the fourth embodiment includes first to fourth time-frequency conversion units 402, 403, 405, and 407 that perform time-frequency conversion. For this reason, the signal is transmitted for each predetermined block length L. Divide and process. In the following description, it is assumed that the block number from the processing start point is represented by k.
- the first time frequency conversion unit 402 performs time frequency conversion of the input signal x (n) for each block length L, and obtains a frequency element X ( ⁇ , k) of the input signal.
- ⁇ is a subscript representing the frequency.
- DFT Discrete Fourier Transform
- the second time frequency conversion unit 403 performs time frequency conversion on the first residual signal d 1 (n) to obtain a frequency element D 1 ( ⁇ , k) of the first residual signal.
- the step size control unit 408 includes the frequency element D 1 ( ⁇ , k) of the first residual signal, the frequency requirement D 2 ( ⁇ , k) of the second residual signal, and the preceding update step size ⁇ prio ( ⁇ , K), the variable update step size ⁇ opt ( ⁇ , k) is calculated for each frequency element.
- the frequency element X ( ⁇ , k) of the input signal is also used for calculation.
- the variable update step size ⁇ opt ( ⁇ , k) can be determined, for example, by the following equation (36).
- variable update step size mu opt corresponding to the state (omega, k) an update step size control section 408 that calculates a calculated variable update step size ⁇ opt ( ⁇ , k) the second performing coefficient updating with Since the update filter 410 is provided, when the disturbance condition differs depending on the frequency band, for example, when there is a frequency band with a large disturbance and a frequency band with a small disturbance, a small value is obtained in the frequency band with a large disturbance.
- the update step size By calculating the update step size, the degradation of the convergence state is suppressed, and a large update step size is calculated in the frequency band where the disturbance is small. Ri identification error can be reduced more quickly.
- the frequency converters 402, 403, 405, 407, the update step size control unit 408, and the second update filter 410 described above are provided, there is sufficient error.
- an appropriate update step size is given to each frequency band, so unnecessary coefficient updating is performed in the frequency band where the error has converged And updating the coefficient in the frequency band where the error convergence is not sufficient.
- FIG. 11 is a block diagram showing the configuration of the adaptive equalizer of the fifth embodiment.
- an adaptive equalizer 500 includes a first subband decomposing unit 501, a second subband decomposing unit 502, and an adaptive unit having the functions described in any of the first to third embodiments.
- An adaptive equalization unit sequence 503 in which a plurality of equalization units are arranged, a first subband synthesis unit 504, and a second subband synthesis unit 505 are configured.
- the adaptive equalizer 500 is connected to the unknown system 900.
- the first subband decomposition unit 501 divides the input signal x (n) to the unknown system 900 into a predetermined number M of frequency bands, and the subband divided input signal x (1) (n) , X (2) (n),..., X (M) (n) are obtained.
- the second subband decomposition unit 502 divides the observation response signal y ′ (n) into M frequency bands, and subband-divided observation response signals y ′ (1) (n), y ′. (2) (n),..., Y ′ (M) (n) is obtained.
- the first and second subband decomposition units 501 that divide the input signal x (n) and the observation response signal y ′ (n) into a predetermined number of divisions. 502 and an adaptive equalization unit sequence 503 configured by a plurality of adaptive equalization units that calculate a variable update step size for each of the band-divided signals and perform coefficient updating using the calculated variable update step size. Even if the disturbance condition and the convergence state of the identification error are different for each frequency band, the identification error can be reduced more quickly and sufficiently.
- FIG. 12 is a block diagram showing a configuration of an acoustic echo canceller apparatus according to the sixth embodiment.
- This acoustic echo canceller apparatus 910 includes an adaptive equalizer 600, and any of the adaptive equalizers described in the first to fifth embodiments can be applied to the adaptive equalizer 600. it can.
- the acoustic echo canceller 910 includes a speaker 901 that outputs a reception signal x (n), a microphone 902 that collects a user's conversation voice and converts it into a sound collection signal y ′ (n), and a subtractor described later. 601.
- the speaker 901 and the microphone 902 constitute an unknown system 900.
- the received signal x (n) is an input signal to the unknown system 900
- the collected sound signal y ′ (n) is an output signal from the unknown system 900. It has become.
- the received voice output from the speaker 901 is picked up by the microphone 902 as an echo.
- the collected sound signal y ′ (n) includes the user's conversation voice and background noise in addition to the echo.
- the conversation voice and the background noise included in the collected sound signal y ′ (n) become disturbances that hinder identification for the adaptive equalizer 600.
- the unknown system 900 as an echo path may suddenly change in its transfer function due to movement of a person or an object in the space around the speaker 901 or the microphone 902.
- the adaptive equalizer 600 efficiently reduces the identification error even when there is a disturbance or a change in the transfer function of the unknown system. Since the coefficient update process is executed, the echo cancellation effect of the acoustic echo canceller can be enhanced.
- the acoustic echo canceller is configured to apply the adaptive equalizer 600 that executes the coefficient updating process for efficiently reducing the identification error.
- the echo cancellation effect of the apparatus can be enhanced.
- FIG. 13 is a block diagram showing the configuration of the active noise control apparatus according to the seventh embodiment.
- a noise source 903 indicates a noise generation source
- a primary path 904 is a sound wave path from the noise source 903 to a target position (the installation position of the error microphone 703 in FIG. 13)
- a secondary path 905 is from the speaker 702. The sound wave path to the target position (the installation position of the error microphone 703 in FIG. 13) is shown.
- Noise generated by the noise source 903 is collected by the reference microphone 701 and becomes a reference signal.
- the reference signal is filtered by a secondary path characteristic filter 706 simulating a transfer function of the secondary path 905, and further phase-inverted by a phase inverter 707 to become an input signal x (n), which is input to the adaptive equalizer 700. Is done.
- noise generated by the noise source 903 is collected by the error microphone 703 through the primary path 904 and input to the adaptive equalizer 700 as an observation response signal y ′ (n).
- the error microphone 703 collects incoming sounds other than the noise emitted from the noise source 903.
- the adaptive equalizer 700 regards the input signal x (n) and the observed response signal y ′ (n) as the unknown system input signal and the observed response signal, respectively, and identifies the transfer function of the unknown system.
- the transfer function of the unknown system is expressed by the following equation (37), where G is the transfer function of the primary path and C is the transfer function of the secondary path.
- the adaptive equalizer 700 When the adaptive equalizer 700 identifies the unknown system described above, if a sound other than the target noise is mixed in the signal collected by the error microphone 703, this becomes a disturbance and prevents the identification, and active noise control. It becomes a factor which reduces the noise reduction effect of an apparatus. Further, when the transfer function of the primary path 904 changes, the adaptive equalizer 700 needs to perform re-identification until the identification error becomes sufficiently small. Furthermore, if the change in the transfer function of the primary path 904 is stationary, the adaptive equalizer must always follow this.
- the transfer function can be accurately identified against disturbance, and Even when a change occurs in this, it is possible to quickly follow, and the noise reduction effect of the active noise control device 920 can be enhanced.
- the adaptive equalizer 700 that calculates the filter coefficient sequence, and the reference signal collected by the reference microphone 701 using the filter coefficient sequence are filtered to obtain the control sound. Since the control sound filter 704 that outputs a signal from the speaker 702 is provided so as to cancel out the control sound and the noise, the noise reduction effect is enhanced even when a disturbance other than the target noise enters the error microphone. Can do.
- the adaptive equalizer 700 that quickly identifies the transfer function by following the change of the transfer function of the primary path is provided, the transfer function of the primary path changes. Even in this case, the noise reduction effect can be enhanced.
- the adaptive equalizer, the acoustic echo canceller apparatus, and the active noise control apparatus can reduce the identification error to a sufficiently small level quickly. It can be used for an adaptive equalizer used for identification, and an acoustic echo canceller device and an active noise control device using this adaptive equalizer.
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Abstract
Description
例えばエコーキャンセラでは、エコー経路の伝達関数を同定することにより、エコーの元となる信号からエコー信号を予測し、エコー信号を消去するために用いられる。また能動騒音制御装置では、騒音経路の伝達関数を同定することにより、騒音源の音響信号から、到来する騒音を予測し、これの逆相となる音響信号を生成することによって騒音を打ち消すために用いられる。
またこのような適用等化器を能動騒音制御装置に提供した場合では、目的の騒音以外の音が外乱となる。例えば送風ダクトにおいて送風ファンの騒音を打消す場合に、ダクト内に設けられた誤差検出マイクで拾われる送風ファン以外の音(別の機械装置の動作音や、近くにいる人の声など)は、適応等化器にとって騒音経路の同定を行う上での外乱となり、騒音低減効果を低減させる要因となる。
式(2)におけるNはフィルタ次数である。またnは時系列を表す添え字であり、初期時点のn=0では、式(2)の係数列には何らかの初期値が与えられる。さらにμは更新ステップサイズを表し、ε(n)は所定の適応アルゴリズムによって与えられるフィルタ係数の更新量を表している。ε(n)の例として、一般によく知られているNLMS(学習同定法)アルゴリズムを用いた場合、次の式(3)で表される。
しかし特許文献1に記載されたエコーキャンセラでは、更新ステップサイズμは同定誤差の所要値に応じて定められるため、所要値の大小に応じて更新ステップサイズも増減するという性質がある。この結果、小さな同定誤差を望むほど更新ステップサイズμの値も小さくなり、これによって係数更新が緩慢になることで、同定誤差の収束により多くの更新回数を要するようになり、時間がかかるようになるという問題が生じる。
このような課題は、例えばエコーキャンセラにおいてエコー経路の伝達関数をすみやかに、あるいは十分精度よく同定することができず、十分なエコー消去効果が得られなくなるという問題を生じる。同様に能動騒音制御装置においても、騒音の伝達経路をすみやかに、あるいは精度よく同定できず、十分な騒音の抑制効果が得られなくなるという問題を生じる。
実施の形態1.
図1は、実施の形態1による適応等化器の構成を示すブロック図である。図1において、適応等化器100は、第1の推定応答信号および第1の残差信号を生成する減算フィルタ101、フィルタ係数列の係数更新を行い、更新したフィルタ係数列を用いて第2の推定応答信号を生成する第1の更新フィルタ102、可変更新ステップサイズを定めるステップサイズ制御部103、フィルタ係数列の係数更新を行う第2の更新フィルタ104、およびフィルタ係数列を記憶するメモリ部105で構成されている。さらに、減算フィルタ101は推定応答信号生成部101aおよび減算器101bを備え、第1の更新フィルタ102はフィルタ処理部102a、乗算器102bおよび減算器102cを備え、第2の更新フィルタ104は係数更新処理部104aおよび乗算器104bを備えている。
また、図1に示すように適応等化器100は未知システム900に接続されている。未知システム900は、適応等化器100の学習同定の対象となるシステムである。
NLMSアルゴリズムを用いる場合、上記式(13)は次の式(14)に書き直される。
式(16)は、係数更新の前後における同定誤差の大きさの差分が更新ステップサイズμの二次関数となることを示しており、右辺が負値となれば同定誤差が減少することを意味している。従って、式(16)の右辺の最小値を与える更新ステップサイズμが、「外乱条件と収束状態の中で、同定誤差を最も減少させる更新ステップサイズ」μoptとなり、これは次の式(17)で与えられる。
0<μ´opt(n)<2μopt(n) ・・・(27)
よって、単に同定誤差が減少することをのみ期待するのであれば、第1の残差信号d1(n)、第2の残差信号d2(n)、および先行更新ステップサイズμprio(n)を元に、所定の手続きによって式(26)に示される範囲のいずれかの値に定めた準可変更新ステップサイズμ´opt(n)を第2の更新フィルタ104に用いるようにしてもよい。本発明の適応等化器は、このような場合も含むものとする。
第3の更新フィルタ106は、フィルタ処理部106a、乗算器106bおよび減算器106cを備え、動作は第1の更新フィルタ102と同様である。第1の更新フィルタ102は第1の先行更新ステップサイズμprio1(n)、第3の更新フィルタ106は第2の先行更新ステップサイズμprio2(n)をそれぞれ用いるものとする。また、第3の更新フィルタ106を追加して設けることにより、ステップサイズ制御部103には、3つの残差信号、すなわち減算フィルタ101からの第1の残差信号d1(n)、第1の更新フィルタ102からの第2の残差信号d2(n)、および第3の更新フィルタ106からの第3の残差信号d3(n)が入力される。
パターン1:
dA(n)=d2(n)、dB(n)=d1(n)、μA=μprio1(n)、μB=0
パターン2:
dA(n)=d3(n)、dB(n)=d2(n)、μA=μprio2(n)、
μB=μprio1(n)
パターン3:
dA(n)=d3(n)、dB(n)=d1(n)、μA=μprio2(n)、μB=0
なお、図3では第3の更新フィルタ106を追加して備え、第1の残差信号d1(n)、第2の残差信号d2(n)、および第3の残差信号d3(n)をステップサイズ制御部103に入力する構成を示したが、追加する更新フィルタの数や、ステップサイズ制御部103に入力する残差信号dの数は適宜変更可能である。
図4(a)は、同定誤差は係数の二乗誤差の和を未知システムの伝達関数のフィルタ係数の二乗値の和で正規化した値で示している。
図4における収束特性曲線(1)は実施の形態1の適応等化器100,110が上述した式(22)に基づいた定めた可変更新ステップサイズμopt(n)を用いた場合の収束特性を示している。これに対して、収束特性曲線(2)は更新ステップサイズを固定値0.05、収束特性曲線(3)は更新ステップサイズを固定値0.1、収束特性曲線(4)は更新ステップサイズを固定値0.2と設定した場合の収束特性を示している。なお、また外乱条件は一定とし、一定レベルの白色雑音を観測応答信号に印加している。
図4(a)に示したグラフから明らかなように、この実施の形態1の適応等化器100,110は、更新ステップサイズにいずれの固定値を用いる場合よりも、同定誤差を速やかに減少させ、かつ、同定誤差を小さな値に収束させる効果があることが分かる。
図4(b)の例では、図4(a)と同様の一定の白色雑音を外乱として印加すると共に、区間Aにおいて突発的な外乱変動として、他の区間よりもやや大きな白色雑音をさらに印加している。また地点Bでは、未知システムの変動を想定して、未知システムの伝達関数を瞬間的に完全に別の伝達関数に変化させている。
上述した実施の形態1では、可変更新ステップサイズμopt(n)が算出された後、算出された可変更新ステップサイズμopt(n)を第2の更新フィルタ104に適用し、フィルタ係数列の係数更新を行う構成を示した。しかし、外乱条件や、未知システム900の伝達関数の変動が比較的小さい場合には、前回算出した可変更新ステップサイズμopt(n-1)と最新の可変更新ステップサイズμopt(n)との値で、大きな差が生じない場合も存在する。例えば、外乱が一定の統計的性質をもつ白色雑音、ピンクノイズ、あるいはそれに類するランダム性信号である場合、係数の更新頻度に対して外乱の変化は十分小さいことが想定できるから、算出される可変更新ステップサイズの値は前回と最新で殆ど変化しないことが予想される。このような場合、前回算出した可変更新ステップサイズμopt(n-1)が次の時点(n)においても有効性が期待できる。そこで、この実施の形態2では、先行更新ステップサイズμprio(n)を前回の動作で得られた可変更新ステップサイズμopt(n-1)とする構成を示す。
実施の形態2の構成を備えることにより、実施の形態1で示した第1の更新フィルタ102に第2の更新フィルタ104の機能を代理させることも可能となり、これによって適応等化器の構成をより簡素にすることができる。
図5において、適応等化器200は、第1の推定応答信号および第1の残差信号を生成する減算フィルタ101、フィルタ係数列の係数更新を行い、更新したフィルタ係数列を用いて第2の推定応答信号を生成する第1の更新フィルタ201、可変更新ステップサイズを定めるステップサイズ制御部103、前回の動作で求めた可変更新ステップサイズを第1の更新フィルタ201に与える遅延処理部202およびフィルタ係数列を記憶するメモリ部105で構成されている。さらに、減算フィルタ101は推定応答信号生成部101aおよび減算器101bを備え、第1の更新フィルタ201は係数更新処理部201a、乗算器201bおよび減算器201cを備えている。また、適応等化器200は未知システム900に接続されている。なお、実施の形態1の図1に示した適応等化器100と同一の構成要素には同一の符号を付して説明を省略する。
上述した実施の形態1では、第1の更新フィルタ102において、入力信号x(n)、および観測応答信号y´(n)から先行更新フィルタ係数列の係数更新を行った後、直ちに同一入力信号x(n)と、観測応答信号y´(n)とを用いて第2の残差信号d2(n)を求め、これを可変更新ステップサイズμopt(n)の算出に用いる構成を示した。
外乱条件や適応等化器の収束状態が周波数帯域毎に異なる場合、高速LMSなどの時間周波数変換を用いる適応フィルタアルゴリズムに本発明の適応等化器を適用し、周波数帯域毎に外乱条件と収束状態に応じた可変更新ステップサイズを算出することで、より効率的な係数更新を行うことが期待できる。そこでこの実施の形態4では、時間周波数変換を用いる適応フィルタアルゴリズムを適用した場合の構成例について述べる。
図9において、適応等化器400は、第1の推定応答信号および第1の残差信号を生成する減算フィルタ401、第1および第2の時間周波数変換部402,403、フィルタ係数列の係数更新を行う第1の更新フィルタ404、第3および第4の時間周波数変換部405,407、可変更新ステップサイズを定めるステップサイズ制御部408、フィルタ係数を更新する第2の更新フィルタ410、減算器406、乗算器409およびフィルタ係数列を記憶するメモリ部411で構成されている。さらに、減算フィルタ401は推定応答信号生成部401aおよび減算器401bを備え、第1の更新フィルタ404はフィルタ処理部404aおよび乗算器404bを備え、第2の更新フィルタ410は係数更新処理部410aを備えている。また、適応等化器400は未知システム900に接続されている。
この実施の形態4の適応等化器400は時間周波数変換を行う第1から第4の時間周波数変換部402,403,405,407を備えており、このため信号を所定のブロック長L毎に分割して処理を行う。以下の説明では、処理開始時点からのブロック番号をkで表すものとして説明を行う。
なお、上記式(36)において、|D1(ω,k)|2は第1の残差信号の周波数要素D1(ω,k)の大きさ、|D2(ω,k)|2は第2の残差信号の周波数要素D2(ω,k)の大きさを表している。このように、周波数要素毎に外乱条件と収束状態に応じた更新ステップサイズを与えることが可能となる。
上述した実施の形態4では、時間周波数変換を用いて周波数帯域毎に可変更新ステップサイズを算出する構成を示したが、この実施の形態5では、高い周波数分解能を必要としない場合に、サブバンドフィルタを用いて、サブバンド毎に外乱条件および収束状態に応じた可変更新ステップサイズを算出する構成を示す。
図11は、実施の形態5の適応等化器の構成を示すブロック図である。図11において、適応等化器500は第1のサブバンド分解部501、第2のサブバンド分解部502、上述した実施の形態1から実施の形態3のいずれかで示した機能を有する適応等化部が複数配置された適応等化部列503、第1のサブバンド合成部504、および第2のサブバンド合成部505で構成されている。また、適応等化器500は未知システム900に接続されている。
第1のサブバンド分解部501は、未知システム900への入力信号x(n)を、所定の分割数M個の周波数帯域に分割し、サブバンド分割された入力信号x(1)(n),x(2)(n),・・・,x(M)(n)を得る。同様に、第2のサブバンド分解部502は、観測応答信号y´(n)をM個の周波数帯域に分割し、サブバンド分割された観測応答信号y´(1)(n),y´(2)(n),・・・,y´(M)(n)を得る。
この実施の形態6では、本発明の適応等化器の音響エコーキャンセラ装置における好適な適用例を示す。図12は、実施の形態6による音響エコーキャンセラ装置の構成を示すブロック図である。
この音響エコーキャンセラ装置910は、適応等化器600を備え、当該適応等化器600には上述した実施の形態1から実施の形態5で説明したいずれかの適応等化器を適用することができる。さらに、音響エコーキャンセラ装置910は、受信信号x(n)を出力するスピーカ901、使用者の会話音声を収音して収音信号y´(n)に変換するマイク902、および後述する減算器601を備える。
騒音源903で発生した騒音は、参照マイク701によって収音され、参照信号となる。参照信号は二次経路905の伝達関数を模擬した二次経路特性フィルタ706にてフィルタリングされ、さらに位相反転器707にて位相反転されて入力信号x(n)となり、適応等化器700に入力される。また、騒音源903で発生した騒音は、一次経路904を通じて誤差マイク703で収音され、観測応答信号y´(n)として適応等化器700に入力される。ただし、誤差マイク703には騒音源903から発した騒音以外の到来音も収音される。
適応等化器700は、入力信号x(n)、観測応答信号y´(n)をそれぞれ未知システムの入力信号と観測応答信号とみなし、同未知システムの伝達関数を同定する。当該未知システムの伝達関数は、一次経路の伝達関数をG、二次経路の伝達関数をCとおいた時、次の式(37)で表される。
Claims (19)
- 可変更新ステップサイズを用いて係数更新処理を行った適応フィルタ係数列を用いて、入力信号をフィルタ処理して出力信号を生成する適応等化器において、
前記可変更新ステップサイズは、
前記適応等化器の前回までの動作により得られた適応フィルタ係数列を用いて得られる第1の残差信号と、
前記適応等化器の前回までの動作により得られた適応フィルタ係数列を任意の先行更新ステップサイズを用いて係数更新を行った先行更新適応フィルタ係数列を用いて得られる第2の残差信号との、
大きさの比あるいは大きさの差に比例して定められることを特徴とする適応等化器。 - 前記第2の残差信号は、前回までの動作により得られた適応フィルタ係数列をそれぞれ異なる複数の任意の先行更新ステップサイズを用いて係数更新を行った複数の先行更新適応フィルタ係数列を用いて得られる複数の残差信号であり、
前記可変更新ステップサイズは、前記第1の残差信号および前記複数の第2の残差信号から選択された少なくとも2つの残差信号の大きさの比あるいは大きさの差に比例して定められることを特徴とする請求項1記載の適応等化器。 - 前記可変更新ステップサイズの決定において、前記先行更新ステップサイズを更なる情報として用いることを特徴とする請求項1記載の適応等化器。
- 前記入力信号は、学習同定対象である未知システムに入力される信号であり、
前記可変更新ステップサイズの決定において、当該入力信号を更なる情報として用いることを特徴とする請求項1記載の適応等化器。 - 前記可変更新ステップサイズは、前記第1の残差信号および/または前記第2の残差信号および前記先行更新ステップサイズにより決定される所定の範囲内のいずれかの値に定められることを特徴とする請求項1記載の適応等化器。
- 所定の信号ブロック長単位で前記係数更新を実行するブロック適応アルゴリズムを用い、前記所定の信号ブロック長単位で前記第1の残差信号および/または前記第2の残差信号それぞれの分散、または平均パワーを求め、
前記可変更新ステップサイズの決定において、前記分散または平均パワーを更なる情報として用いることを特徴とする請求項1記載の適応等化器。 - 前記先行更新ステップサイズは、前回までの動作により得られた可変更新ステップサイズであることを特徴とする請求項1記載の適応等化器。
- 前記可変更新ステップサイズは、周波数変換された前記第1の残差信号および/または前記第2の残差信号および前記先行更新ステップサイズの周波数要素毎に定められることを特徴とする請求項3記載の適応等化器。
- 前回までの動作により得られた適応フィルタ係数列を用いて前記入力信号をフィルタ処理して第1の推定応答信号を生成し、前記入力信号に対する前記未知システムの応答信号に外乱が加えられた観測応答信号から、前記生成した第1の推定応答信号を減算して前記第1の残差信号を生成する減算フィルタと、
前記減算フィルタで生成された第1の残差信号および前記任意の先行更新ステップサイズを用いて、前記適応等化器の前回までの動作により得られた適応フィルタ係数列の係数更新を行って先行更新フィルタ係数列を算出し、算出した先行更新フィルタ係数列を用いて前記入力信号をフィルタ処理して第2の推定応答信号を生成し、前記観測応答信号から前記第2の推定応答信号を減算して前記第2の残差信号を生成する更新フィルタと、
少なくとも前記減算フィルタが生成した第1の残差信号、前記更新フィルタが生成した第2の残差信号および前記先行更新ステップサイズとから前記可変更新ステップサイズを定めるステップサイズ制御部とを備えたことを特徴とする請求項1記載の適応等化器。 - 前記更新フィルタは、複数の前記任意の先行更新ステップサイズを用いて算出した複数の先行更新フィルタ係数列を用いて、前記入力信号をフィルタ処理して複数の前記第2の推定応答信号を生成し、前記観測応答信号から前記複数の第2の推定応答信号を減算して複数の第2の残差信号を生成し、
前記ステップサイズ制御部は、少なくとも前記第1の残差信号および前記複数の第2の残差信号から選択された少なくとも2つの残差信号と、前記先行更新ステップサイズとから前記可変更新ステップサイズを定めることを特徴とする請求項9記載の適応等化器。 - 前記可変更新ステップサイズを用いて前記適応フィルタ係数列の係数更新を行う係数更新フィルタを備えたことを特徴とする請求項9記載の適応等化器。
- 前回までの動作により得られた適応フィルタ係数列を用いて前記入力信号をフィルタ処理して第1の推定応答信号を生成し、前記入力信号に対する前記未知システムの応答信号に外乱が加えられた観測応答信号から、前記生成した第1の推定応答信号を減算して前記第1の残差信号を生成する第1の減算フィルタと、
前回までの動作により得られた先行更新フィルタ係数列を用いて前記入力信号をフィルタ処理して第2の推定応答信号を生成し、前記観測応答信号から、前記生成した第2の推定応答信号を減算して前記第2の残差信号を生成する第2の減算フィルタと、
少なくとも前記第1の残差信号、前記第2の残差信号および前記先行更新ステップサイズとから前記可変更新ステップサイズを定めるステップサイズ制御部と、
前回までの動作により得られた係数更新量と前記ステップサイズ制御部が定めた可変更新ステップサイズを元に、前記適応フィルタ係数列を更新し、前記入力信号を更新した適応フィルタ係数列によりフィルタ処理して出力推定応答信号を生成し、生成した出力推定応答信号を前記観測応答信号から減算して出力残差信号を求める係数更新フィルタと、
前記入力信号および前記係数更新フィルタが求めた出力残差信号から前記係数更新量を算出する係数更新量算出部と、
前記係数更新フィルタが更新した適応フィルタ係数列を元に、前記係数更新量算出部が算出した係数更新量、および前記先行更新ステップサイズとから前記先行更新フィルタ係数列を得る先行係数更新部とを備えたことを特徴とする請求項1記載の適応等化器。 - 時間変化する前記入力信号、前記第1の残差信号および前記第2の残差信号を周波数変換して周波数要素を得る時間周波数変換部を備え、
前記ステップサイズ制御部は、前記時間周波数変換部において周波数変換された少なくとも前記第1の残差信号および/または前記第2の残差信号および前記先行更新ステップサイズの周波数要素を元に、周波数要素毎に可変更新ステップサイズを定めることを特徴とする請求項9記載の適応等化器。 - 時間変化する前記入力信号、前記第1の残差信号および前記第2の残差信号を周波数変換して周波数要素を得る時間周波数変換部を備え、
前記ステップサイズ制御部は、前記時間周波数変換部において周波数変換された少なくとも前記第1の残差信号および/または前記第2の残差信号および前記先行更新ステップサイズの周波数要素を元に、周波数要素毎に可変更新ステップサイズを定めることを特徴とする請求項12記載の適応等化器。
- 可変更新ステップサイズを用いて係数更新処理を行った適応フィルタ係数列を用いて、入力信号をフィルタ処理して出力信号を生成する適応等化器において、
前記入力信号は学習同定対象である未知システムに入力される信号であり、当該入力信号と、当該入力信号に対する前記未知システムの応答信号に外乱が加えられた観測応答信号とをサブバンドに分解するサブバンド分解部と、
前記適応等化器の前回までの動作により得られた適応フィルタ係数列を用いて得られる第1の残差信号と、
前記適応等化器の前回までの動作により得られた適応フィルタ係数列を任意の先行更新ステップサイズを用いて係数更新を行った先行更新適応フィルタ係数列を用いて得られる第2の残差信号との、
大きさの比あるいは大きさの差に比例して前記可変更新ステップサイズを定め、
定めた可変更新ステップサイズを元に前記適応フィルタ係数列を更新し、更新された適応フィルタ係数列を用いて前記サブバンド分解部により分解された入力信号をフィルタ処理して出力推定信号を生成し、生成した出力推定信号を前記観測応答信号から減算して出力残差信号を求める適応等化部と、
前記適応等化部がサブバンド毎に求めた前記出力残差信号をサブバンド合成するサブバンド合成部とを備えたことを特徴とする適応等化器。 - 前記入力信号および前記観測応答信号を元に、前記入力信号が入力される学習同定対象である未知システムの伝達関数を同定し、前記未知システムの応答信号を推定した出力推定応答信号を出力する請求項1記載の適応等化器と、
前記観測応答信号から前記適応等化器が出力する出力推定応答信号を減算し、送信信号を出力する減算器とを備えたことを特徴とする音響エコーキャンセラ装置。 - 前記入力信号および前記観測応答信号を元に、前記入力信号が入力される学習同定対象である未知システムの伝達関数を同定し、前記未知システムの応答信号を推定した出力推定応答信号を出力する請求項15記載の適応等化器と、
前記観測応答信号から前記適応等化器が出力する出力推定応答信号を減算し、送信信号を出力する減算器とを備えたことを特徴とする音響エコーキャンセラ装置。 - 騒音を集音した参照信号に対してフィルタ処理および位相反転処理を行った前記入力信号および前記観測応答信号を元に、前記入力信号が入力される学習同定対象である未知システムの伝達関数を同定する請求項1記載の適応等化器と、
前記適応等化器が同定した前記未知システムの伝達関数を示すフィルタ係数列を用いて前記参照信号をフィルタ処理し、制御音信号を生成する制御音フィルタと、
前記制御音フィルタが生成した制御音信号を出力するスピーカとを備え、
前記騒音を、前記スピーカから出力された前記制御音信号で相殺することを特徴とする能動騒音制御装置。 - 騒音を集音した参照信号に対してフィルタ処理および位相反転処理を行った前記入力信号および前記観測応答信号を元に、前記入力信号が入力される学習同定対象である未知システムの伝達関数を同定する請求項15記載の適応等化器と、
前記適応等化器が同定した前記未知システムの伝達関数を示すフィルタ係数列を用いて前記参照信号をフィルタ処理し、制御音信号を生成する制御音フィルタと、
前記制御音フィルタが生成した制御音信号を出力するスピーカとを備え、
前記騒音を、前記スピーカから出力された前記制御音信号で相殺することを特徴とする能動騒音制御装置。
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- 2012-03-29 JP JP2013513902A patent/JP5496418B2/ja not_active Expired - Fee Related
- 2012-03-29 CN CN201280013679.7A patent/CN103444094B/zh not_active Expired - Fee Related
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JP2014174349A (ja) * | 2013-03-08 | 2014-09-22 | Toshiba Corp | 能動消音装置および能動消音方法 |
CN103208281A (zh) * | 2013-03-12 | 2013-07-17 | 武汉大学 | 一种变步长调整的变压器噪声有源控制方法 |
WO2016021114A1 (ja) * | 2014-08-05 | 2016-02-11 | パナソニックIpマネジメント株式会社 | 信号処理装置、プログラム、レンジフード装置 |
JP2016038416A (ja) * | 2014-08-05 | 2016-03-22 | パナソニックIpマネジメント株式会社 | 信号処理装置、プログラム、レンジフード装置 |
US10229666B2 (en) | 2014-08-05 | 2019-03-12 | Panasonic Intellectual Property Management Co., Ltd. | Signal processing device, program, and range hood device |
WO2019044176A1 (ja) * | 2017-08-28 | 2019-03-07 | ソニー株式会社 | 音声処理装置及び音声処理方法、並びに情報処理装置 |
JPWO2019044176A1 (ja) * | 2017-08-28 | 2020-10-01 | ソニー株式会社 | 音声処理装置及び音声処理方法、並びに情報処理装置 |
US11245983B2 (en) | 2017-08-28 | 2022-02-08 | Sony Corporation | Audio processing device and method for echo cancellation |
US20230298559A1 (en) * | 2022-03-17 | 2023-09-21 | Airoha Technology Corp. | Adaptive active noise control system with unstable state handling and associated method |
US11942068B2 (en) * | 2022-03-17 | 2024-03-26 | Airoha Technology Corp. | Adaptive active noise control system with unstable state handling and associated method |
Also Published As
Publication number | Publication date |
---|---|
CN103444094A (zh) | 2013-12-11 |
US20130315408A1 (en) | 2013-11-28 |
EP2675073A1 (en) | 2013-12-18 |
JPWO2012153451A1 (ja) | 2014-07-31 |
EP2675073A4 (en) | 2015-07-22 |
CN103444094B (zh) | 2016-06-08 |
JP5496418B2 (ja) | 2014-05-21 |
EP2675073B1 (en) | 2018-01-10 |
US9830900B2 (en) | 2017-11-28 |
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