JP6614534B2 - Signal processing device, program, and range hood device - Google Patents

Description

  The present invention relates to a signal processing device, a program, and a range hood device.

  Conventionally, there is a silencer using active noise control as a technique for reducing noise in a space (noise propagation path) through which sound emitted from a noise source propagates. Active noise control is a technique for actively reducing noise by radiating a cancellation sound having the opposite phase and the same amplitude.

  For example, there is a configuration in which a canceling sound is generated by updating a filter coefficient of a silencer filter using an LMS algorithm (LMS: Least Mean Square). And in order to suppress a residual noise, suppressing the gain increase of the muffler filter in a specific frequency band, patent document 1 discloses a configuration using white noise. Specifically, in the LMS algorithm, the silencer includes a first reference signal generated from the output of the reference microphone, a second reference signal generated from the white noise, a first error signal generated from the output of the error microphone, The filter coefficient is updated using the second error signal generated from the white noise.

  However, the problem that the gain of the muffler filter increases in the low frequency band has not been sufficiently solved even with the configuration of Patent Document 1.

JP-A-8-22292

  The objective of this invention is providing the signal processing apparatus, program, and range hood apparatus which can fully suppress the gain of the muffler filter in a low frequency band.

  A signal processing apparatus according to an aspect of the present invention includes a first sound input device that is provided in a target space in which noise emitted from a noise source propagates, collects the noise, and receives the cancellation signal to input the noise. Used in combination with a sound input / output device including a sound output device that emits a canceling sound to be canceled to the target space, and a second sound input device that collects a synthesized sound of the noise and the canceling sound in the target space. A mute filter that sets a filter coefficient and outputs the cancel signal based on the output of the first sound input device, and a first that blocks a frequency component less than a threshold frequency from the output of the first sound input device. From a first filter that outputs a reference signal, a first signal processing unit that generates a second reference signal having a uniform power spectrum below the threshold frequency, and an output of the second sound input device A second filter that outputs a first error signal that cuts off a frequency component less than the threshold frequency, and a second error signal that is a signal less than the threshold frequency based on the second reference signal and the filter coefficient. A signal processing unit, and a coefficient updating unit that calculates the filter coefficient based on the first reference signal, the second reference signal, the first error signal, and the second error signal are provided.

  A program according to one embodiment of the present invention causes a computer to function as the above-described signal processing device.

  A range hood device according to one aspect of the present invention is directed to the above-described signal processing device, the sound input / output device, a hollow cylindrical air passage that forms the target space, and one end of the air passage toward the other end. And an air blower that generates an air flow.

It is a block diagram which shows the structure of the silencer of Embodiment 1. It is a perspective view which shows the external appearance of the range hood apparatus of Embodiment 1. FIG. FIG. 3A is a graph showing noise characteristics of the first embodiment. FIG. 3B is a graph showing a silencing characteristic by noise suppression processing different from the present embodiment. FIG. 6 is a block diagram illustrating a configuration of a first signal processing unit in a first modification of the first embodiment. It is a block diagram which shows the structure of the silencer in the 2nd modification of Embodiment 1. FIG. 10 is an explanatory diagram illustrating interrupt processing in a second modification of the first embodiment. It is a block diagram which shows the structure of the silencer of Embodiment 2. It is the schematic which shows the relationship of the some duct of Embodiment 2. FIG.

  Hereinafter, embodiments of the present invention will be described with reference to the drawings.

  The following embodiments generally relate to signal processing devices, programs, and range hood devices. More specifically, the following embodiments relate to a signal processing device, a program, and a range hood device for performing active noise control.

(Embodiment 1)
FIG. 1 shows a configuration of a silencer 1 (active noise control device) of the present embodiment, and the range hood device 2 includes the silencer 1.

  As shown in FIG. 2, the range hood apparatus 2 includes a duct 21 (ventilation path) disposed above a kitchen appliance in the kitchen. The duct 21 is formed in a box shape having an opening on the lower surface, and a current plate 23 is provided in the opening. The rectifying plate 23 is formed slightly smaller than the opening. The air inlet 21 a is formed around the rectifying plate 23. Furthermore, the duct 21 includes a fan 22 (see FIG. 1) that takes in indoor air into the duct 21 from the air inlet 21a and exhausts it outside the room. An operation unit 24 is provided on the front surface of the range hood device 2. The operation unit 24 includes an operation switch for each operation of the range hood device 2, an indicator lamp indicating an operation state, and the like. In addition, the space in the duct 21 constituting the air passage corresponds to a space where noise propagates.

  When the fan 22 operates, the fan 22 becomes a noise source, and the operation sound (noise) of the fan 22 propagates through the duct 21 and is transmitted from the air inlet 21a to the room. Therefore, the silencer 1 is provided in the duct 21 in order to suppress noise transmitted to the room during the operation of the fan 22.

  The silencer 1 provided in the duct 21 includes a sound input / output device 11 and a silencer control device 12 as shown in FIG.

  The sound input / output device 11 includes a reference microphone 111 (first sound input device), an error microphone 112 (second sound input device), and a speaker 113 (sound output device). The reference microphone 111 is located on the fan 22 side in the duct 21. The error microphone 112 is located on the inlet 21 a side in the duct 21. The speaker 113 is located between the reference microphone 111 and the error microphone 112 in the duct 21. That is, the reference microphone 111, the speaker 113, and the error microphone 112 are arranged in this order from the fan 22 to the air inlet 21a.

  The mute control device 12 includes amplifiers 121, 122, 123, A / D converters 124, 125, a D / A converter 126, and a signal processing device 13.

  The output of the reference microphone 111 is amplified by the amplifier 121 and then A / D converted by the A / D converter 124. The output of the A / D converter 124 is input to the signal processing device 13.

  The output of the error microphone 112 is amplified by the amplifier 122 and then A / D converted by the A / D converter 125. The output of the A / D converter 125 is input to the signal processing device 13.

  The cancel signal output from the signal processing device 13 is D / A converted by the D / A converter 126 and then amplified by the amplifier 123. The speaker 113 receives the cancel signal amplified by the amplifier 123 and emits a cancel sound.

  The signal processing device 13 is configured by a computer that executes a program. Then, the signal processing device 13 causes the speaker 113 to output a cancel sound that cancels the noise generated by the fan 22 so that the sound pressure level at the installation point (silence point) of the error microphone 112 is minimized. That is, when the speaker 113 outputs a canceling sound, noise transmitted from the fan 22 to the outside of the duct 21 through the air inlet 21a is suppressed. The signal processing device 13 performs active noise control. The signal processing device 13 executes a muffling program that realizes the function of the adaptive filter in order to follow the noise change and noise propagation characteristic change of the fan 22 as a noise source. For updating the filter coefficient of the adaptive filter, a Filtered-X LMS (Least Mean Square) sequential update control algorithm (hereinafter referred to as a Filtered-X LMS algorithm) is used.

  Hereinafter, the operation of the signal processing device 13 will be described.

  First, the reference microphone 111 collects noise from the fan 22 and outputs a noise signal including the collected noise to the mute control device 12. The A / D converter 124 performs A / D conversion on the noise signal amplified by the amplifier 121 at a predetermined sampling frequency. The A / D converter 124 outputs the A / D converted discrete value to the signal processing device 13.

  The error microphone 112 collects residual noise that could not be erased by the cancellation sound at the silencing point, and outputs an error signal corresponding to the collected residual noise to the silencing control device 12. The A / D converter 125 A / D converts the error signal amplified by the amplifier 122 at the same sampling frequency as the A / D converter 124. The A / D converter 125 outputs the A / D converted discrete value to the signal processing device 13 as a time domain error signal e (t).

  The signal processing device 13 includes a howling cancellation filter 131 (Howling Cancel Filter), a subtraction unit 132, a correction filter 133, a low cut filter 134 (Low Cut Filter) (first filter), a first addition unit 135, and a low cut filter 136 (second filter). Filter), a second addition unit 137, a first signal processing unit 138, a second signal processing unit 139, a coefficient updating unit 140, and a mute filter 141. The low cut filter 134 and the low cut filter 136 are hereinafter referred to as LCF 134 and LCF 136.

  Further, the first signal processing unit 138 includes a noise generation unit 138a and a low pass filter 138b (Low Pass Filter). The second signal processing unit 139 includes a filtering processing unit 139a, a multiplication unit 139b, and a subtraction unit 139c. The coefficient update unit 140 includes a coefficient calculation unit 140a, converters 140b and 140c, and inverse converters 140d and 140e. Hereinafter, the low-pass filter 138b is referred to as LPF 138b.

  The howling cancellation filter 131 is an FIR filter (Finite Impulse Response Filter). The howling cancel filter 131 is set with a transfer characteristic F ^ simulating the transfer characteristic F of sound waves from the speaker 113 to the reference microphone 111 as a filter coefficient. Note that the transfer characteristic simulating the transfer characteristic F is represented by a symbol F ^ in which F is a mountain-shaped symbol ^ (hat symbol). Further, in this specification, the symbol ^ is arranged obliquely above F, and the symbol ^ is arranged directly above F in FIGS. Represents a characteristic.

  This howling cancellation filter 131 performs a convolution operation on the transfer characteristic F ^ on the cancellation signal Y (t) output from the muffler filter 141. Then, the subtractor 132 outputs a signal obtained by subtracting the output of the howling cancellation filter 131 from the output of the A / D converter 124. That is, a signal obtained by subtracting the wraparound component of the canceling sound from the noise signal collected by the reference microphone 111 is output from the subtracting unit 132 as the noise signal X (t). Therefore, even if the cancel sound emitted from the speaker 113 wraps around the reference microphone 111, occurrence of howling can be prevented. The output of the subtraction unit 132 is input to the mute filter 141 and the correction filter 133.

  The muffler filter 141 is an FIR type adaptive filter. The silencing filter 141 is set with the filter coefficient W (t) by the coefficient updating unit 140. In the silence filter 141 of the present embodiment, filter coefficients W1 (t) to Wn (t) are set for each of a plurality of frequency bins obtained by dividing the entire frequency band of the cancel sound into n. In addition, when not distinguishing the filter coefficients W1 (t) to Wn (t) in the time domain, they are represented as filter coefficients W (t). Further, the number n of frequency bins is set so that the frequency width of the frequency bin is, for example, several tens Hz to several hundreds Hz.

  The correction filter 133 is an FIR filter. In the correction filter 133, a transfer characteristic C ^ simulating the transfer characteristic C of a sound wave from the speaker 113 to the error microphone 112 is set as a filter coefficient. The transfer characteristic simulating the transfer characteristic C is represented by a symbol C ^ with a mountain-shaped symbol ^ appended to C. Further, in this specification, the symbol ^ is arranged diagonally above C, and the symbol ^ is arranged directly above C in FIGS. 1, 5, and 7, but in each case, the transmission simulating the transfer characteristic C is performed. Represents a characteristic.

  Then, the correction filter 133 performs a convolution operation between the noise signal X (t) output from the subtraction unit 132 and the transfer characteristic C ^. The output of the correction filter 133 is input to the LCF 134 as a time domain reference signal r (t). The LCF 134 outputs a first reference signal r1 (t) obtained by blocking a low frequency band less than the threshold frequency f1 from the reference signal r (t). In the present embodiment, the frequency band of the noise to be canceled by the cancellation sound (silence target band) is determined in advance (for example, several tens Hz to several kHz), and the threshold frequency f1 is the lower limit value of the silence target band.

  In addition, the first signal processing unit 138 generates a second reference signal r2 (t) in which a band equal to or higher than the threshold frequency f1 is blocked and a power spectrum at a frequency lower than the threshold frequency f1 is uniform. Specifically, the noise generation unit 138a generates a broadband white noise signal and outputs it to the LPF 138b. The LPF 138b outputs a second reference signal r2 (t) obtained by blocking a band having a threshold frequency f1 or higher from a broadband white noise signal. The second reference signal r2 (t) is input to the first adder 135 and the second signal processor 139.

  The first adding unit 135 adds the first reference signal r1 (t) and the second reference signal r2 (t), and outputs the addition result to the coefficient updating unit 140 as a combined reference signal r3 (t).

  The error signal e (t) is input to the LCF 136. The LCF 136 outputs a first error signal e1 (t) obtained by blocking a low frequency band less than the threshold frequency f1 from the error signal e (t).

  In the second signal processing unit 139, the filtering processing unit 139a is an FIR type adaptive filter. In the filtering processing unit 139a, the coefficient updating unit 140 sets the same filter coefficient W (t) as that of the silence filter 141. In other words, the filtering coefficients W1 (t) to Wn (t) are also set for the plurality of frequency bins obtained by dividing the entire frequency band of the canceling sound into n in the filtering processing unit 139a. Then, the filtering processor 139a performs a convolution operation on the second reference signal r2 (t) and outputs the filter coefficient W (t) (filtering process).

  Further, in the second signal processing unit 139, the multiplication unit 139b multiplies the second reference signal r2 (t) by the target gain and outputs the result. The target gain is a target value of the gain of the muffler filter 141 in the low frequency band lower than the threshold frequency f1, and is set to “0.1” if the amplitude ratio of the input to the output is, for example. When this target gain is expressed in common logarithm, it is −20 dB (= 20 · log 0.1). The target gain is a preset fixed value or a value that can be arbitrarily changed. The smaller the target gain, the smaller the gain of the muffler filter 141 in the low frequency band below the threshold frequency f1.

  In the second signal processing unit 139, the subtraction unit 139c generates the second error signal e2 (t) by subtracting the output of the filtering processing unit 139a from the output of the multiplication unit 139b. The second error signal e2 (t) represents the amount by which the gain in the low frequency band (less than the threshold frequency f1) by the filtering process of the filter coefficient W (t) is shifted from the target gain.

  The second addition unit 137 adds the first error signal e1 (t) and the second error signal e2 (t), and outputs the addition result to the coefficient updating unit 140 as a combined error signal e3 (t). The combined error signal e3 (t) is a wideband signal that combines the first error signal e1 (t) and the second error signal e2 (t).

  In the coefficient updating unit 140, the converter 140b converts the combined reference signal r3 (t) in the time domain into a combined reference signal R (ω) in the frequency domain by FFT (Fast Fourier Transform). The converter 140c converts the combined error signal e3 (t) in the time domain into a combined error signal E (ω) in the frequency domain by FFT.

  The coefficient calculation unit 140a of the coefficient update unit 140 calculates the filter coefficients W1 (ω) to Wn (ω) of the mute filter 141 and the filtering processing unit 139a using the Filtered-X LMS algorithm in the frequency domain. The coefficient calculation unit 140a receives the combined reference signal R (ω) and the combined error signal E (ω), and calculates filter coefficients W1 (ω) to Wn (ω). In addition, when not distinguishing the filter coefficients W1 (ω) to Wn (ω) in the frequency domain, they are represented as filter coefficients W (ω).

  In general, in the process of updating the filter coefficient W (ω) using the frequency-domain Filtered-X LMS algorithm, the filter coefficient W (ω) is obtained so that the combined error signal E (ω) is minimized. Specifically, the update process of the filter coefficient W (ω) is expressed by [Equation 1] where W (ω) is the filter coefficient, μ is the update parameter, and m is the sample number. The update parameter μ is also called a step size parameter. The update parameter μ is a parameter that determines the magnitude of the correction amount of the filter coefficient W (ω) in the process of repeatedly calculating the filter coefficient W (ω) using the LMS algorithm or the like.

  Each of the inverse transformers 140d and 140e performs inverse FFT (Inverse Fast Fourier Transform), thereby converting the frequency domain filter coefficients W1 (ω) to Wn (ω) calculated by the coefficient calculation unit 140a into time domain filters. The coefficients W1 (t) to Wn (t) are converted. Filter coefficients W1 (t) to Wn (t) for each frequency bin of the muffler filter 141 are set by the output of the inverse converter 140d. Filter coefficients W1 (t) to Wn (t) for each frequency bin of the filtering processing unit 139a are set by the output of the inverse converter 140e. In FIG. 1, two inverse transformers 140d and 140e are provided. However, the filter coefficients of the mute filter 141 and the filtering processing unit 139a may be set by the output of one inverse transformer.

  As described above, the coefficient update unit 140 sequentially updates the filter coefficients W1 (t) to Wn (t). The filtering processing unit 139a separates the second reference signal r2 (t) for each frequency bin. The filtering processing unit 139a performs a convolution operation between the second reference signal r2 (t) and the filter coefficient W (t) for each frequency bin. Then, the filtering processing unit 139a outputs the sum of the results of the convolution operation performed for each frequency bin to the subtraction unit 139c. The subtractor 139c generates the second error signal e2 (t) by subtracting the output of the filtering processor 139a from the output of the multiplier 139b. The second addition unit 137 adds the first error signal e1 (t) and the second error signal e2 (t), and outputs the addition result to the coefficient updating unit 140 as a combined error signal e3 (t).

  Further, the mute filter 141 separates the noise signal X (t) for each frequency bin. The silencing filter 141 performs a convolution operation between the noise signal X (t) and the filter coefficient W (t) for each frequency bin. Then, the silence filter 141 outputs the sum of the results of the convolution operation performed for each frequency bin as a cancel signal Y (t). The cancel signal Y (t) output from the muffler filter 141 is D / A converted by the D / A converter 126 and then amplified by the amplifier 123, and a cancel sound is output from the speaker 113.

  The waveform of the cancellation sound (cancellation signal Y (t)) is generated so as to have the opposite phase and the same amplitude as the noise waveform at the silencing point, and is propagated from the fan 22 through the duct 21 and discharged from the intake port 21a. Noise is reduced.

  Here, in the first reference signal r1 (t), the low frequency band lower than the threshold frequency f1 is cut off, and the frequency component equal to or higher than the threshold frequency f1 remains mainly. The second reference signal r2 (t) is a signal in a low frequency band below the threshold frequency f1 based on white noise without dip and notch. Therefore, the synthesized reference signal r3 (t) is a signal based on white noise having no dip and notch in the frequency band below the threshold frequency f1, and the frequency band equal to or higher than the threshold frequency f1 is a signal based on the output of the reference microphone 111. Become.

  The first error signal e1 (t) has a low frequency band lower than the threshold frequency f1 cut off, and mainly has a frequency component equal to or higher than the threshold frequency f1. The second error signal e2 (t) is a signal in a low frequency band below the threshold frequency f1 based on white noise having no dip and notch. Therefore, the combined error signal e3 (t) is a signal based on white noise having no dip and notch in the frequency band below the threshold frequency f1, and the frequency band equal to or higher than the threshold frequency f1 is a signal based on the output of the error microphone 112. Become.

  That is, the synthesized reference signal r3 (t) and the synthesized error signal e3 (t) have flat frequency characteristics in which the low frequency band below the threshold frequency f1 has no dip and notch. Therefore, the filter coefficient W (t) is obtained based on the combined reference signal r3 (t) and the combined error signal e3 (t) having such frequency characteristics. As a result, the gain of the muffler filter 141 in the low frequency band below the threshold frequency f1 is sufficiently suppressed.

  FIG. 3A shows a silencing characteristic by the signal processing device 13 of the present embodiment. A characteristic Y1 (solid line) indicates the sound pressure (amplitude) at the mute point (the installation point of the error microphone 112) when the noise suppression process of the present embodiment is performed. Further, the characteristic Y0 (broken line) indicates the sound pressure (amplitude) at the silence point when the noise suppression process is not performed.

  FIG. 3B shows noise suppression processing by the Filtered-X LMS algorithm, which is noise suppression processing different from that of the present embodiment, and which receives the reference signal r (t) and the error signal e (t). A characteristic Y11 (solid line) indicates a sound pressure (amplitude) at a muffle point when a noise suppression process different from the present embodiment is performed. A characteristic Y10 (broken line) indicates the sound pressure (amplitude) at the muffle point when the noise suppression process is not performed.

  In FIG. 3B, in the low frequency band below the threshold frequency f1, the sound pressure of the characteristic Y11 in the low frequency band is amplified by the increase in the gain of the muffler filter 141, and is higher than the sound pressure of the characteristic Y10. (Refer area | region R10 in FIG. 3B).

  On the other hand, in FIG. 3A, the gain of the muffler filter 141 is sufficiently suppressed in a low frequency band lower than the threshold frequency f1. Therefore, the sound pressure of the characteristic Y1 in the low frequency band is suppressed and is almost the same as the sound pressure of the characteristic Y0 (see region R1 in FIG. 3A). That is, it can be seen that the signal processing device 13 of the present embodiment can suppress the oscillation in the low frequency band by sufficiently suppressing the gain of the silence filter 141 in the low frequency band lower than the silence target band.

  As a first modification of the present embodiment, a first signal processing unit 138A shown in FIG. 4 can be provided instead of the first signal processing unit 138. The first signal processing unit 138A stores the data of the second reference signal r2 (t) in advance, and reads out the data of the second reference signal r2 (t) from the storage unit 138c to obtain the second reference signal r2. A signal output unit 130d for outputting (t).

  The storage unit 138c is a memory such as a ROM (Read Only Memory). The storage unit 138c stores in advance data of the second reference signal r2 (t) obtained by blocking a band of the threshold frequency f1 or higher from the broadband white noise signal. Then, the signal output unit 130d repeatedly reads out the data of the second reference signal r2 (t) from the storage unit 138c and outputs the second reference signal r2 (t).

  Therefore, the first signal processing unit 138A can reduce the load required for the generation process of the second reference signal r2 (t) compared to the first signal processing unit 138.

  As a second modification of the present embodiment, the signal processing device 13 preferably includes a buffer 142 as shown in FIG. The buffer 142 is provided between the subtraction unit 139c and the second addition unit 137. The buffer 142 stores the discrete value of the second error signal e2 (t) output from the subtracting unit 139c by a first-in first-out (FIFO) method.

  The signal processing device 13 is composed of a computer that executes a program, and in this case, interrupt processing as shown in FIG. 6 is performed.

  The signal processing device 13 includes, as the interrupted normal process J1, a first process S1 including a filter coefficient update process, a second reference signal r2 (t) generation process, and a second error signal e2 (t) generation process. Then, the second process S2 including the writing process of the buffer 142 is performed. The normal process J1 is a process that does not require real-time performance.

  Further, the signal processing device 13 performs the third process S11 to the seventh process S15 as the interrupt process J2 that interrupts the normal process J1. In the third process S <b> 11, the first stored discrete value is transferred to the coefficient updating unit 140 from the second error signal e <b> 2 (t) stored in the buffer 142 through the addition process of the second addition unit 137. The fourth process S12 is a filtering process of the noise signal X (t) by the silencing filter 141 (convolution operation of the noise signal X (t) and the filter coefficient W (t)). The fifth process S13 is an anti-disturbance process that stops the filter coefficient update process when a disturbance sound is input. The sixth process S14 is a filtering process performed by the LCFs 134 and 136. The seventh process S15 is an addition process by the first addition unit 135 and the second addition unit 137. This interrupt process J2 is a process executed for each sample of the noise signal X (t) and the error signal e (t) obtained by A / D converting the outputs of the reference microphone 111 and the error microphone 112, and is real time. This is a process that requires high performance.

  Basically, a process that does not require real-time property is preferably processed in the normal process J1. For example, the filtering process (third process S11) of the noise signal X (t) by the mute filter 141 is a process that is necessarily required for each sample, and thus is processed by the interrupt process J2. However, if the amount of processing in the interrupt process J2 becomes too large, the filtering process (S12) by the muffler filter 141 is not properly performed, or other processes (S13 to S15) that require real-time performance are executed with high accuracy. It may not be possible.

  Therefore, in this modification, the normal process J1 includes the second process S2 including the process of generating the second reference signal r2 (t), the process of generating the second error signal e2 (t), and the process of writing the buffer 142. The amount of processing in the interrupt processing J2 is reduced. When the interrupt process J2 is started, the discrete values in the buffer 142 are transferred to the coefficient update unit 140 through the addition process of the second adder 137, and each process of the interrupt process J2 is performed. The buffer 142 writes and reads discrete values using a first-in first-out method.

(Embodiment 2)
The structure of the silencer 1 of this embodiment is shown in FIG. In the silencer 1, the same components as those in the first embodiment are denoted by the same reference numerals, and description thereof is omitted.

  The silencer 1 includes a signal processing device 13A, and the signal processing device 13A includes a bandpass filter 134A (first filter) and a bandpass filter 136A (second filter). Further, the first signal processing unit 138A includes a notch filter 138e. Hereinafter, the band-pass filter 134A, the band-pass filter 136A, and the notch filter 138e are referred to as BPF 134A, BPF 136A, and BEF (Band Elimination Filter) 138e. BPF134A, BPF136A, and BEF138e correspond to the LCF134, LCF136, and LPF138b of the first embodiment.

  As shown in FIG. 8, the ducts 21 of the plurality of range hood devices 2, 2 </ b> A may be adjacent to each other. In this case, oscillation may occur due to crosstalk (mutual interference) between the ducts 21. Specifically, it is known that oscillation easily occurs when the following conditions (1) and (2) are satisfied.

  (1) In the range hood apparatus 2 to be silenced, there is a notch band that is a frequency band in which the gain of the transfer characteristic C (the transfer characteristic of the sound wave from the speaker 113 to the error microphone 112) falls locally.

  (2) The sound (path D in FIG. 8) that circulates from the speaker 113 of the range hood apparatus 2A adjacent to the range hood apparatus 2 to the error microphone 112 of the range hood apparatus 2 has a large sound pressure in the above-described notch band. .

  That is, when the conditions (1) and (2) are satisfied at the same time, the range hood device 2 is likely to oscillate in the notch band.

  Therefore, the present embodiment aims to suppress oscillation due to this notch band when a notch band exists in a non-target band that is a frequency band other than the muffing target band.

  The non-target band is a frequency band that does not require silencing control, and includes a frequency band where the sound wave propagating in the duct 21 does not become a plane wave (depending on the diameter of the duct 21), or a band where the power of the noise spectrum is small. .

  The frequency band in which the sound wave propagating in the duct 21 does not become a plane wave is mainly a high frequency band. In this high frequency band, even if the sound pressure level at the silencing point is minimized, the sound pressure level may be amplified outside the duct 21 away from the silencing point, so that silencing control is unnecessary. The frequency fo (Hz) that does not become a plane wave is generally determined by the diameter L (m) of the duct 21, and “fo ≧ C / 2L” when the sound speed is C (m / s). That is, the plane wave is not generated at the lower limit frequency fo1 (= C / 2L) or more.

  Specifically, the signal processing device 13A of the present embodiment has the following configuration.

  The BPF 134A sets a passband that is greater than or equal to the threshold frequency f1 (first threshold frequency) and less than the threshold frequency f2 (second threshold frequency) (f1 <f2 ≦ fo1). Therefore, the first reference signal r1 (t) mainly includes frequency components of the reference signal r (t) that are equal to or higher than the threshold frequency f1 and lower than the threshold frequency f2, and are frequency components that are lower than the threshold frequency f1 and higher than or equal to the threshold frequency f2. Becomes a blocked signal. In this case, the threshold frequency f1 is a lower limit value of the silence target band, and the threshold frequency f2 is an upper limit value of the silence target band. The threshold frequency f2 is a frequency equal to or lower than the lower limit frequency fo1.

  The BPF 136A sets a passband between the threshold frequency f1 and the threshold frequency f2. Therefore, the first error signal e1 (t) mainly includes frequency components of the error signal e (t) that are equal to or higher than the threshold frequency f1 and lower than the threshold frequency f2, and are frequency components that are lower than the threshold frequency f1 and higher than or equal to the threshold frequency f2. Becomes a blocked signal.

  The BEF 138e uses a passband that is less than the threshold frequency f1 and greater than or equal to the threshold frequency f2. Therefore, the second reference signal r2 (t) becomes white noise at a frequency lower than the threshold frequency f1 and higher than the threshold frequency f2, and is a signal in which a frequency component higher than the threshold frequency f1 and lower than the threshold frequency f2 is blocked.

  That is, the first reference signal r1 (t) and the first error signal e1 (t) are actually collected signals, and are signals in a band not lower than the threshold frequency f1 and lower than the threshold frequency f2. In addition, the second reference signal r2 (t) and the second error signal e2 (t) are a low frequency band less than the threshold frequency f1 (a non-target band lower than the silence target band), and a high frequency band (the silence target) higher than the threshold frequency f2. (Non-target band higher than the band) and is generated based on white noise. In other words, the second reference signal r2 (t) and the second error signal e2 (t) mainly contain white noise components having no dip and notch in each band below the threshold frequency f1 and above the threshold frequency f2. The actually collected signal is hardly included.

  That is, the synthesized reference signal r3 (t) and the synthesized error signal e3 (t) have flat frequency characteristics with no dip and notch in the low frequency band below the threshold frequency f1 and the high frequency band above the threshold frequency f2. Therefore, since the filter coefficient W (t) is obtained based on the synthesized reference signal r3 (t) and the synthesized error signal e3 (t) having such frequency characteristics, the silence filter in the low frequency band below the threshold frequency f1. The gain of 141 is sufficiently suppressed. Furthermore, oscillation due to crosstalk (mutual interference) between the ducts 21 in a high frequency band equal to or higher than the threshold frequency f2 can be suppressed.

  Further, the configuration of the second embodiment can be combined with the configurations of the first and second modifications of the first embodiment, and the same effects as those of the first and second modifications of the first embodiment can be obtained.

  As is clear from the embodiment described above, the signal processing device 13 or 13A according to the first aspect of the present invention includes a reference microphone 111 (first sound input device), a speaker 113 (sound output device), and an error. It is used in combination with a sound input / output device 11 including a microphone 112 (second sound input device). The reference microphone 111 is provided in a target space (space in the duct 21) through which noise emitted from the fan 22 (noise source) propagates, and collects noise. The speaker 113 receives a cancel signal and emits a cancel sound that cancels the noise to the target space. The error microphone 112 collects a synthesized sound of noise and cancellation sound in the target space.

  The signal processing device 13 includes a mute filter 141, an LCF 134 (first filter), a first signal processing unit 138, an LCF 136 (second filter), a second signal processing unit 139, and a coefficient updating unit 140. . The mute filter 141 is set with a filter coefficient and outputs a cancel signal Y (t) based on the output of the reference microphone 111. The LCF 134 outputs a first reference signal r1 (t) obtained by blocking a frequency component less than the threshold frequency f1 from the output of the reference microphone 111. The first signal processing unit 138 generates a second reference signal r2 (t) in which the power spectrum at a frequency lower than the threshold frequency f1 is uniform. The LCF 136 outputs a first error signal e1 (t) obtained by blocking a frequency component less than the threshold frequency f1 from the output of the error microphone 112. The second signal processing unit 139 generates a second error signal e2 (t) that is a signal less than the threshold frequency f1 based on the second reference signal r2 (t) and the filter coefficient. The coefficient updating unit 140 calculates a filter coefficient based on the first reference signal r1 (t), the second reference signal r2 (t), the first error signal e1 (t), and the second error signal e2 (t).

  In the present embodiment, the LCF 134 generates the first reference signal r1 (t) in which the low frequency band below the threshold frequency f1 is cut off. Further, the LCF 136 generates a first error signal e1 (t) that cuts off a low frequency band less than the threshold frequency f1. In addition, the first signal processing unit 138 creates a second reference signal r2 (t) that cuts off the band of the threshold frequency f1 or more from white noise. Also, the second signal processing unit 139 creates a second error signal e2 (t) that cuts off the band of the threshold frequency f1 or more from white noise.

  That is, the first reference signal r1 (t) and the first error signal e1 (t) are actually collected signals in a band equal to or higher than the threshold frequency f1. Further, the second reference signal r2 (t) and the second error signal e2 (t) are signals in a low frequency band (a non-target band lower than a frequency band that is a target of mute control) that is less than the threshold frequency f1 based on white noise. It is. That is, the second reference signal r2 (t) and the second error signal e2 (t) mainly contain white noise components having a dip and notch and less than the threshold frequency f1, and the actually collected signals are It is hardly included. Therefore, the coefficient updating unit 140 sets the first reference signal r1 (t), the second reference signal r2 (t), the first error signal e1 (t), and the second error signal e2 (t). The filter coefficient W (t) can sufficiently suppress the gain of the muffler filter 141 in the low frequency band below the threshold frequency f1.

  In the signal processing device 13 or 13A according to the second aspect of the present invention, in the first aspect, the second signal processing unit 139 includes a filtering processing unit 139a, a multiplication unit 139b, and a subtraction unit 139c. It is preferable. The filtering processor 139a sets the filter coefficient, filters the second reference signal r2 (t), and outputs the filtered signal. The multiplier 139b multiplies the second reference signal r2 (t) by a predetermined target gain. The subtractor 139c generates the second error signal e2 (t) by subtracting the output of the filtering processor 139a from the output of the multiplier 139b.

  Therefore, the second error signal e2 (t) represents an amount by which the gain in the low frequency band (less than the threshold frequency f1) by the filtering process using the filter coefficient W (t) is deviated from the target gain. it can.

  Moreover, it is preferable that the signal processing device 13 or 13A according to the third aspect of the present invention further includes the first addition unit 135 and the second addition unit 137 in the first or second aspect. The first adder 135 adds the first reference signal r1 (t) and the second reference signal r2 (t). The second adder 137 adds the first error signal e1 (t) and the second error signal e2 (t). Coefficient updating unit 140 calculates a filter coefficient based on the output of first addition unit 135 and the output of second addition unit 137.

  Therefore, the coefficient updating unit 140 can easily perform the filter coefficient calculation process using the Filtered-X LMS algorithm.

  The signal processing device 13 or 13A according to the fourth aspect of the present invention preferably includes the first signal processing unit 138A in any one of the first to third aspects. The first signal processing unit 138A stores the data of the second reference signal r2 (t) in advance, and reads out the data of the second reference signal r2 (t) from the storage unit 138c to obtain the second reference signal r2. A signal output unit 130d for outputting (t).

  Therefore, the first signal processing unit 138A can reduce the load required for the generation process of the second reference signal r2 (t) compared to the first signal processing unit 138.

  Further, in the signal processing device 13 or 13A according to the fifth aspect of the present invention, in any one of the first to fourth aspects, the second signal processing unit 139 generates the second error signal e2 (t). Output discrete values. The signal processing device 13 further includes a buffer 142 that sequentially stores each of a plurality of discrete values of the second error signal e2 (t) output from the second signal processing unit 139. Then, as an interrupt process interrupting each process of the first signal processing unit 138 and the second signal processing unit 139, a plurality of discrete values of the second error signal e2 (t) stored in the buffer 142 are coefficients in a first-in first-out method. It is preferable that processing delivered to the update unit 140 is performed.

  Therefore, it is possible to appropriately perform a filtering process by the muffler filter 141 that requires real-time performance, and to improve the muffling performance.

  In addition, in the signal processing device 13 or 13A according to the sixth aspect of the present invention, in any one of the first to fifth aspects, the coefficient updating unit 140 performs a filter coefficient calculation process in the frequency domain. Is preferred.

  In this case, since the calculation amount of the filter coefficient in the frequency domain is small compared to the calculation process of the filter coefficient in the time domain, the load on the coefficient updating unit 140 can be reduced.

  In addition, in the signal processing device 13 or 13A according to the seventh aspect of the present invention, in any one of the first to sixth aspects, the threshold frequency f1 is a frequency band of noise to be canceled by the canceling sound. The lower limit value is preferable.

  Therefore, the gain of the silencing filter 141 in the low frequency band lower than the silencing target band is sufficiently suppressed, so that the influence on the actual silencing performance can be suppressed.

  In the signal processing device 13 or 13A according to the eighth aspect of the present invention, in any one of the first to seventh aspects, the BPF 134A receives the threshold frequency f1 (first threshold value) from the output of the reference microphone 111. Frequency) and a band pass filter that passes a frequency component that is higher than the threshold frequency f1 and lower than the threshold frequency f2 (second threshold frequency) is preferable. The BPF 136A is a band-pass filter that passes a frequency component equal to or higher than the threshold frequency f1 and lower than the threshold frequency f2 from the output of the error microphone 112. The first signal processing unit 138A generates a second reference signal r2 (t) in which the power spectrum at each of the frequencies below the threshold frequency f1 and above the threshold frequency f2 is uniform.

  Therefore, the gain of the silencing filter 141 in the frequency band outside the silencing target band (low frequency band, high frequency band) is sufficiently suppressed, and oscillation due to crosstalk (mutual interference) between the ducts 21 outside the silencing target band is also suppressed. be able to.

  In each of the above-described embodiments, the computer included in the signal processing devices 13 and 13A includes a processor and an interface that operate according to a program as main hardware configurations. Such processors include a DSP (Digital Signal Processor), a CPU (Central Processing Unit), an MPU (Micro-Processing Unit), and the like. And if a processor can implement | achieve each function of the above-mentioned signal processing apparatuses 13 and 13A by running a program, the kind will not ask | require.

  As a program providing form, a computer-readable ROM (Read Only Memory), a form stored in advance in a recording medium such as an optical disc, or the like is supplied to the recording medium via a wide area communication network including the Internet. There are forms.

  That is, the program according to the ninth aspect of the present invention causes a computer to function as the signal processing device 13 or 13A.

  Therefore, a program that causes a computer to function as the signal processing device 13 or 13A can also achieve the same effect as described above. That is, this program can sufficiently suppress the gain of the muffler filter 141 in the low frequency band below the threshold frequency f1.

  Moreover, the range hood apparatus 2 of the 10th aspect which concerns on this invention is the signal processing apparatus 13 or 13A, the sound input / output apparatus 11, the hollow cylindrical duct 21 (ventilation path) which comprises object space, and a duct. And a fan 22 (blower device) that generates an air flow from one end to the other end.

  Therefore, the range hood apparatus 2 equipped with the signal processing apparatuses 13 and 13A can sufficiently suppress the gain of the silence filter 141 in the low frequency band lower than the threshold frequency f1.

  Moreover, devices other than the range hood device 2 may include the muffler device 1 of each of the above-described embodiments.

  The above-described embodiment is an example of the present invention. For this reason, the present invention is not limited to the above-described embodiment, and various modifications can be made depending on the design and the like as long as the technical idea according to the present invention is not deviated from this embodiment. Of course, it is possible to change.

DESCRIPTION OF SYMBOLS 1 Silencer 11 Sound input / output device 12 Silencer control device 13, 13A Signal processing device 111 Reference microphone (1st sound input device)
112 Error microphone (second sound input device)
113 Speaker (sound output device)
131 Howling Cancel Filter 132 Subtraction Unit 133 Correction Filter 134, 134A Low Cut Filter (First Filter)
135 1st addition part 136,136A Low cut filter (2nd filter)
137 2nd addition part 138 1st signal processing part 139 2nd signal processing part 139a Filtering processing part 139b Multiplication part 139c Subtraction part 140 Coefficient update part 141 Silencer filter 2 Range hood apparatus 21 Duct (air passage)
22 fans

Claims (10)

A first sound input device that collects the noise provided in a target space through which noise generated from a noise source propagates, and a sound output device that generates a cancel sound that cancels the noise when a cancel signal is input to the target space And a sound input / output device including a second sound input device that collects a synthesized sound of the noise and the cancellation sound in the target space,
A mute filter that is set with a filter coefficient and outputs the cancel signal based on the output of the first sound input device;
A first filter that outputs a first reference signal obtained by blocking a frequency component less than a threshold frequency from the output of the first sound input device;
A first signal processing unit that generates a second reference signal having a uniform power spectrum below the threshold frequency;
A second filter for outputting a first error signal obtained by blocking a frequency component lower than the threshold frequency from the output of the second sound input device;
A second signal processing unit that generates a second error signal that is a signal less than the threshold frequency based on the second reference signal and the filter coefficient;
And a coefficient updating unit that calculates the filter coefficient based on the first reference signal, the second reference signal, the first error signal, and the second error signal. The second signal processor is
A filtering processor configured to filter and output the second reference signal, the filter coefficient being set;
A multiplier for multiplying the second reference signal by a predetermined target gain;
The signal processing apparatus according to claim 1, further comprising: a subtracting unit that generates the second error signal by subtracting the output of the filtering processing unit from the output of the multiplying unit. A first adder for adding the first reference signal and the second reference signal; and a second adder for adding the first error signal and the second error signal;
The signal processing apparatus according to claim 1, wherein the coefficient updating unit calculates the filter coefficient based on an output from the first addition unit and an output from the second addition unit.   The first signal processing unit includes a storage unit that stores data of the second reference signal in advance, and a signal output unit that reads the data of the second reference signal from the storage unit and outputs the second reference signal. The signal processing apparatus according to claim 1, further comprising: The second signal processing unit outputs a discrete value of the second error signal;
A buffer that sequentially stores each of a plurality of discrete values of the second error signal output by the second signal processing unit;
As an interrupt process interrupting each process of the first signal processing unit and the second signal processing unit, a plurality of discrete values of the second error signal stored in the buffer are delivered to the coefficient updating unit in a first-in first-out manner. The signal processing apparatus according to claim 1, wherein the processing is performed.   The signal processing apparatus according to claim 1, wherein the coefficient updating unit performs the filter coefficient calculation process in a frequency domain.   The signal processing apparatus according to claim 1, wherein the threshold frequency is a lower limit value of a frequency band of the noise to be canceled by the cancel sound. The first filter is a band-pass filter that passes from the output of the first sound input device a frequency component that is equal to or higher than a first threshold frequency and lower than a second threshold frequency that is higher than the first threshold frequency;
The second filter is a band-pass filter that passes a frequency component that is equal to or higher than the first threshold frequency and lower than the second threshold frequency from the output of the second sound input device,
The said 1st signal processing part produces | generates the 2nd reference signal from which the power spectrum in each less than the said 1st threshold frequency and above the 2nd threshold frequency becomes uniform. The signal processing device according to one item.   A program for causing a computer to function as the signal processing apparatus according to any one of claims 1 to 8.   The signal processing device according to any one of claims 1 to 8, the sound input / output device, a hollow cylindrical air passage constituting the target space, and one end of the air passage toward the other end. A range hood device comprising a blower that generates an airflow.

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