JP5338769B2 - Coefficient setting method of digital filter, coefficient setting device, coefficient setting program, and sound field correction method using digital filter - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a method, device and program for coefficient setting and a sound field correction method that can set a coefficient array of a digital filter for performing sound field correction with a small amount of calculation. <P>SOLUTION: The method includes: a step S11 for sampling the signal value of a signal output by a microphone that collects measurement sounds; a step S04 for setting a first coefficient of a digital filter that outputs an audio signal to a speaker that outputs the measurement sounds, as a reference signal value; and a step S05 for setting an additive inverse of a signal value that is sampled at the time different from the time at which the signal value set as the first coefficient is sampled by prescribed length of time (that is, a corrected distortion signal) as a second coefficient to be used for arithmetic processing of an audio signal of the time different from the time of the audio signal on which the digital filter performs arithmetic processing with the use of the first coefficient, by prescribed length of time. <P>COPYRIGHT: (C)2012,JPO&amp;INPIT

Description

  The present invention relates to a coefficient setting method for a digital filter, a coefficient setting apparatus, a coefficient setting program, and a sound field correction method using a digital filter.

  Patent Document 1 discloses a technique for performing sound field correction using a digital filter. That is, Patent Document 1 discloses a technique for correcting an audio signal with a digital filter so as to cancel noise and sound distortion generated in a sound field. In this technique, a predetermined correction is performed on a converted value obtained by performing a fast Fourier transform on the impulse response of the microphone, and a value obtained by performing an inverse Fourier transform on the converted converted value is set as a coefficient of the digital filter.

JP-A-8-33092

  The technique according to Patent Document 1 has a problem that the amount of calculation is large because it is necessary to perform fast Fourier transform and inverse Fourier transform in order to calculate the filter coefficient of the digital filter.

  The present invention has been made in view of the above problems, and provides a coefficient setting method, a coefficient setting apparatus, a coefficient setting program, and a digital filter that can set the coefficient array of a digital filter that performs sound field correction with a small amount of calculation. An object of the present invention is to provide a sound field correction method used.

In order to achieve the above object, a digital filter coefficient setting method according to a first aspect of the present invention includes:
A sampling step of sampling the signal value of the signal output from the microphone that collected the measurement sound;
A first setting step of setting a first coefficient of a digital filter that outputs an audio signal to the speaker that outputs the measurement sound, to the signal value sampled in the sampling step;
A second coefficient used for arithmetic processing on the audio signal at a time different from the audio signal that the digital filter performs arithmetic processing using the first coefficient by a predetermined time is used as the first coefficient in the first setting step. A second setting step of setting to a reciprocal of the signal value sampled at a time different from the sampling time of the signal value set to the predetermined time;
It is characterized by having.

In the digital filter according to the first aspect,
A reference signal value selection step of selecting a reference signal value to be set in the first coefficient from the sampled signal value based on the magnitude of the signal value sampled in the sampling step;
Based on the reference signal value selected in the reference signal value selection step and the sampled signal value, the distortion signal value representing the distortion of the sound output from the speaker is sampled. A distortion signal value selection step for selecting from the signal value obtained,
The first setting step sets the reference signal value selected in the reference signal value selection step to the first coefficient,
The second setting step sets a reciprocal of the distortion signal value selected in the distortion signal value selection step to the second coefficient;
It is also good.

Furthermore, in the digital filter according to the first aspect,
In the sampling step, when the speaker receives an impulse signal, the response signal value output from the microphone is sampled,
In the reference signal value selection step, the largest signal value among the response signal values sampled in the sampling step is selected as the reference signal value,
In the distortion signal value selection step, the signal value smaller than the reference signal value is selected as the distortion signal value among the response signal values sampled in the sampling step.
It is also good.

Furthermore, in the digital filter according to the first aspect,
A signal value storage step of storing in the storage device a plurality of correspondence information in which the response signal value sampled in the sampling step and the sampling time information indicating the sampling time of the response signal value are associated;
An average value calculating step of averaging the response signal value stored in the storage device in association with the sampling time information in the signal value storing step based on the sampling time information;
In the reference signal value selection step, the largest value among the response signal values averaged in the average value calculation step is selected as the reference signal value,
In the distortion signal value selection step, a value smaller than the reference signal value is selected as the distortion signal value among the response signal values averaged in the average value calculation step.
It is also good.

In order to achieve the above object, a coefficient setting device for a digital filter according to a second aspect of the present invention includes:
Sampling means for sampling the signal value of the signal output from the microphone that collected the measurement sound;
First setting means for setting a first coefficient of a digital filter that outputs an audio signal to the speaker that outputs the measurement sound, to the signal value sampled by the sampling means;
A second coefficient used for arithmetic processing for the audio signal at a time different from the audio signal calculated by the digital filter using the first coefficient by a predetermined time is used as the first coefficient by the first setting means. Second setting means for setting to a reciprocal of the signal value sampled at a time different from the sampling time of the signal value set to the predetermined time;
It is characterized by providing.

In order to achieve the above object, a coefficient setting program for a digital filter according to a third aspect of the present invention includes:
Computer
Sampling means for sampling the signal value of the signal output from the microphone collecting the measurement sound,
First setting means for setting, to the signal value sampled by the sampling means, a first coefficient of a digital filter that outputs an audio signal to a speaker that outputs the measurement sound;
A second coefficient used for arithmetic processing on the audio signal at a time different from the audio signal calculated by the digital filter using the first coefficient by a predetermined time is calculated by the first setting means by the first coefficient. Second setting means for setting a reciprocal of the signal value sampled at a time different from the sampling time of the signal value set by the predetermined time;
It is made to function as.

In order to achieve the above object, a sound field correction method using a digital filter according to a fourth aspect of the present invention includes:
A sampling step of sampling the signal value of the signal output from the microphone that collected the measurement sound;
A first setting step of setting a first coefficient of a digital filter that outputs an audio signal to the speaker that outputs the measurement sound, to the signal value sampled in the sampling step;
A second coefficient used for arithmetic processing on the audio signal at a time different from the audio signal that the digital filter performs arithmetic processing using the first coefficient by a predetermined time is used as the first coefficient in the first setting step. A second setting step of setting to a reciprocal of the signal value sampled at a time different from the sampling time of the signal value set to the predetermined time;
The digital filter performs an arithmetic processing on the audio signal using the first coefficient whose value is set in the first setting step and the second coefficient whose value is set in the second setting step to the speaker. Output audio signal output step;
It is characterized by having.

  According to the present invention, the coefficient array of the digital filter that performs sound field correction can be set with a small amount of calculation.

FIG. 1A is a system configuration diagram illustrating an example of an audio system including a coefficient setting device used in a sound field correction method and a coefficient setting method according to an embodiment of the present invention. FIG. 1B is a hardware configuration diagram illustrating a configuration example of the coefficient setting device. It is a figure showing the example of 1 structure of a digital filter. FIG. 3A is a flowchart illustrating an example of coefficient setting processing executed by the coefficient setting device. FIG. 3B is a flowchart illustrating an example of a response signal acquisition process executed by the coefficient setting device. FIG. 3C is a flowchart illustrating an example of correction processing executed by the coefficient setting device. FIG. 4A is a functional block diagram illustrating an example of functions that the coefficient setting device has. FIG. 4B is a diagram illustrating a configuration example of the response signal acquisition unit. FIG. 4C is a diagram illustrating a configuration example of the correction unit. FIG. 5A is a diagram illustrating an example of the average sampling signal. FIG. 5B is a diagram illustrating an example of a table in which sampling times are associated with average sampling signal magnitudes. FIG. 5C is a diagram illustrating an example of a coefficient array set in the digital filter. FIG. 6A is a diagram illustrating an example of an impulse response output from the digital filter in which the coefficient is set. FIG. 6B is a diagram illustrating an example of the impulse response of the speaker.

  Hereinafter, an audio system including a filter coefficient setting device used for implementing a filter coefficient setting method and a sound field correction method according to an embodiment of the present invention will be described with reference to the accompanying drawings.

  The audio system 100 according to the present embodiment is a car audio system that performs sound field correction on a digital audio signal output from a playback device and emits sound into a vehicle compartment that is a sound field. As shown in FIG. 1A, the audio system 100 includes a player device 10, a sound field correction device 20, an output circuit 30, a speaker 40, and a microphone 50, and is mounted on a vehicle.

  The player device 10 is a playback device that reads a digital audio signal from a recording medium and outputs the read signal. The recording medium includes, for example, a CD (Compact Disc), a DVD (Digital Versatile Disc), a Blu-ray disc, a hard disk, and a flash memory.

  The sound field correction device 20 includes a digital filter 21 and a filter coefficient setting device (hereinafter referred to as coefficient setting device) 22. The sound field correction device 20 is a kind of noise canceller that corrects an audio signal in advance so as to cancel distortion and noise generated in the sound output from the audio system 100 to the sound field. Note that audio distortion, noise, and the like are not included in the signal reproduced from the player device 10 such as abnormal sound caused by abnormal vibration of the speaker 40, reverberation sound generated in the sound field, and the like. It means sound distortion or abnormal sound caused by the sound field itself.

The digital filter 21 performs a filter process on the digital audio signal output from the audio device 10 according to the set filter coefficient, and outputs the processed signal to the speaker 40.
The filter coefficient setting device 22 sets a filter coefficient for canceling distortion and noise generated in the sound output to the sound field in the digital filter 21.

The output circuit 30 includes a DA (Digital / Analog) converter, an output amplifier, and the like, converts the digital audio signal output from the sound field correction device 20 into an analog audio signal, and outputs the converted signal to a speaker.
The speaker 40 converts the output signal of the output circuit 30 into air vibration and outputs it to the passenger compartment (sound field).
The microphone 50 is installed at the listening position at the time of measuring distortion and noise of the sound output to the sound field. At the time of measurement, the microphone 50 acquires the sound output from the speaker 40, converts the acquired sound into an electrical signal, and outputs the converted signal. The microphone 50 is detached from the audio system 100 during normal use.

  The sound field refers to a space or place where sound exists and includes the interior of the vehicle. In addition, sound distortion and noise generated in the sound field include reverberant sound. Specifically, since a speaker 40 that outputs sound is installed in a space or place where the sound exists, sound distortion and noise generated in the sound field are vibrations generated when the speaker 40 outputs sound. Including audio distortion due to That is, sound distortion and noise generated in the sound field vary depending on the vehicle type or the type of the speaker 40.

  Next, the configuration of the digital filter 21 shown in FIG. The digital filter 21 is configured by, for example, an FIR (Finite Impulse Response) filter, and the filter coefficient is set by the coefficient setting device 22. In addition, the digital filter 21 performs arithmetic processing on a digital audio signal (hereinafter simply referred to as an audio signal) output from the player device 10 during normal use, using already set filter coefficients, and processed audio. The signal is output to the output circuit 30. This is because the noise signal included in the audio signal is filtered.

  As shown in FIG. 2, the digital filter 21 is composed of six taps connected in order from the input side to which the audio signal X is input to the output side to output the audio signal Y. Here, the tap located closest to the input side is designated as the 0th tap, and the remaining five taps are sequentially named in the same manner. Each tap includes a delay element, a multiplier, and an adder. The delay element delays the input signal by one sampling period (that is, one sampling period), and then multiplies the delayed signal by the same tap and the adjacent tap located on the output side by one. Output to. The multiplier multiplies the signal delayed by the delay element by the coefficient value set by the coefficient setting device 22. The adder multiplies the signal output from the adjacent tap located on the input side by one (that is, the tap for processing a signal whose delay time is shorter by one sampling period) by the coefficient value by the multiplier. Add the received signals. After that, the adder outputs the added signal to the adjacent tap located on the output side by one (that is, a tap for processing a signal having a delay time longer by one sampling period).

  Here, the coefficient of the multiplier included in the 0th tap is denoted as K0, and the coefficients of the multipliers included in the remaining five taps are sequentially named K1 to K5. Further, the arrangement of the coefficient values of the multipliers respectively provided in the 0th tap to the 5th tap in order from K0 is simply referred to as a coefficient arrangement of the digital filter 21 hereinafter. Each element of the coefficient array will be described as being arranged in order from the coefficient used for processing the audio signal with a short delay time, but the present invention is not limited to this.

Next, the coefficient setting device 22 for setting the coefficient of the digital filter 21 will be described.
As shown in FIG. 1B, the coefficient setting device 22 includes a central processing unit (CPU) 22a, a random access memory (RAM) 22b, a read only memory (ROM) 22c, an input unit 22d, and an output unit 22e. Is done.
The CPU 22a performs overall processing of the coefficient setting device 22 by executing software processing according to a program stored in the ROM 22c. The RAM 22b is a storage device that temporarily stores data to be processed by the CPU 22a.
The input unit 22d includes, for example, an AD (Analog / Digital) converter, inputs an analog signal output from the microphone 50, and converts the input signal into a digital signal. The output unit 22e is controlled by the CPU 22a and outputs a measurement signal to be described later to the output circuit 30. The output unit 22e is controlled by the CPU 22a to set the coefficient of the digital filter 21 in the digital filter 21.

  The CPU 22a in FIG. 1B executes a coefficient setting process as shown in FIG. 3A, whereby a measurement signal output unit 221, a response signal acquisition unit 222, and a correction unit as shown in FIG. 4A. 223 and the setting unit 225, and in cooperation with the RAM 22b, functions as a coefficient array database (hereinafter simply referred to as coefficient DB) 224.

  When the execution of the coefficient setting process of FIG. 3A is started, the measurement signal output unit 221 of FIG. 4A is an impulse signal (hereinafter referred to as a measurement signal) used for measuring audio distortion and noise. Is output to the speaker 40 via the output circuit 30 (step S01).

  Next, the response signal acquisition unit 222 executes a response signal acquisition process as shown in FIG. 3B for acquiring a response signal representing a response to the measurement signal (step S02). As shown in FIG. 4B, the response signal acquisition unit 222 includes a sampling unit 222a, a signal value storage unit 222b, a signal value database (hereinafter referred to as signal value DB) 222c, and an averaging unit 222d. .

  When the execution of the response signal acquisition process illustrated in FIG. 3B is started, the sampling unit 222a in FIG. 4B performs a signal value of the response signal output from the microphone 50 (hereinafter simply referred to as a response signal value). Are sampled (that is, sampled) at the sampling period of the digital filter 21 (step S11).

  Thereafter, the signal value storage unit 222b converts the signal value sampled by the sampling unit 222a (hereinafter referred to as the sampled signal value) into sampling time information indicating the sampling time (that is, the sampling time), and the number of measurements. Correspondence information associated with the information representing the information is generated, and the generated correspondence information is stored in the signal value DB 222c (step S12). The sampling time is measured based on the measurement start time that is the time when the measurement signal output unit 221 outputs the measurement signal. According to these configurations, even when the response signal is measured for each measurement signal output over a plurality of times, the value of the response signal at each time can be compared for each sampling time.

  After step S12 is executed, the averaging unit 222d averages the sampling signal values accumulated in the signal value DB 222c in association with the sampling time information by the signal value storage unit 222b based on the sampling time information. (Step S13). As a specific example, the averaging unit 222d calculates an average value of a plurality of sampled signal values sampled at the same sampling time. Thereafter, the response signal acquisition unit 222 ends the execution of the response signal acquisition process. According to this configuration, for example, an error in signal value caused by noise can be reduced.

  After step S02 of FIG. 3A is executed, the correction unit 223 of FIG. 4A changes the coefficient value of the digital filter 21 to a coefficient value for canceling audio distortion or noise based on the measurement result. Correction processing as shown in FIG. 3C is executed (step S03). As shown in FIG. 4C, the correction unit 223 includes a reference signal value selection unit 223a, a distortion signal value selection unit 223b, a coefficient array generation unit 223c, an inverse correction unit 223d, a signal value storage unit 223e, and a normalization. Unit 223f and coefficient array storage unit 223g.

  When the execution of the correction process shown in FIG. 3C is started, the reference signal value selection unit 223a performs the sampling signal value averaged in step S13 in FIG. 3B (hereinafter, the average sampling signal value). The reference signal value is selected (step S21). Here, the reference signal value refers to a signal value used for determining an average sampled signal value (hereinafter referred to as a distorted signal value) representing distortion or noise of the measured voice among the average sampled signal value. Specifically, the reference signal value selection unit 223a selects the largest average sampling signal value as the reference signal value. Usually, the impulse response is because a plurality of distortion spectra having a signal value smaller than the signal value are generated after the largest signal value.

  Here, as shown in FIGS. 5A and 5B, the magnitude (level) of the average sampling signal value at the time of the first sample is equal to the average sampling signal at the time of the second to sixth samples. Since the value is larger than the value, the reference signal value selection unit 223a selects the average sampling signal value of the first sample as the reference signal value. In FIG. 5A, the horizontal axis represents time, and the vertical axis represents the average response signal value (level). FIG. 5B is a table that displays the sampling time and the average sampling signal value in association with each other.

  After step S21 in FIG. 3C is executed, the distortion signal value selection unit 223b in FIG. 4C performs the magnitude of the reference signal value selected by the reference signal value selection unit 223a and the average sampling signal value. One or a plurality of distortion signal values are selected from the average sampled signal values based on the magnitude of (step S22). Usually, when the voice is distorted, the volume of the voice is distorted, and according to this configuration, the distortion signal value can be accurately selected.

  More specifically, the distortion signal value selection unit 223b determines an average sampling signal value smaller than the reference signal value as the distortion signal value. That is, as shown in FIGS. 5A and 5B, the average sampled signal values after the second sample from the level “1.0” of the average sampled signal value at the time of the first sample as the reference signal value. Therefore, the distortion signal value selection unit 223b determines the average sampling signal value after the second sample as the distortion signal value. This is because the volume of distorted audio is usually smaller than the volume of undistorted audio.

  After step S22 of FIG. 3C is executed, the coefficient array generation unit 223c of FIG. 4C generates a coefficient array of the digital filter 21 as shown in FIG. 5C (step S23). . The coefficient array in FIG. 5C has element numbers A0 to A5. In the element identified by the element number A0, the coefficient value set to the coefficient K0 of the digital filter 21 shown in FIG. 2 is stored. Similarly, the coefficient values set in the coefficients K1 to K5 are stored in the respective elements identified by the element numbers A1 to A5.

  After step S23 in FIG. 3C is executed, the inverse correction unit 223d in FIG. 4C reverses the sign of the distortion signal value and corrects it to the inverse of the value before correction. Execute (step S24). This is because sound distortion and noise are canceled out using the inverse of the distortion signal value. Note that a certain reciprocal number is a number in which a value obtained by adding the certain number to the value “0”.

  Next, the signal value storage unit 223e and the reference signal value selected by the reference signal value selection unit 223a and the distortion signal value whose sign is reversed by the inverse correction unit 223d (hereinafter simply referred to as a corrected distortion signal). Are stored in the coefficient array in ascending order of sampling time of the reference signal value and the corrected distortion signal (step S25). This is because the elements of the coefficient array are arranged in order from the coefficients used for processing an audio signal with a short delay time.

  After step S25 of FIG. 3C is executed, the normalization unit 223f of FIG. 4C converts the element of the coefficient array stored by the signal value storage unit 223e into the signal value ( That is, normalization is performed so that the magnitude of the reference signal value is 1 (step S26). As a specific example, the normalization unit 223f divides each element of the coefficient array by the absolute value of the reference signal value.

  Next, the coefficient array storage unit 223g stores the coefficient array whose values are stored by the signal value storage unit 223e in the coefficient DB 224 (step S27). Thereafter, the correction unit 223 in FIG. 4B ends the execution of the correction process.

  After the execution of step S03 illustrated in FIG. 3A is completed, the first setting unit 225a included in the setting unit 225 illustrated in FIG. 4A includes the reference signal at the position A0 of the coefficient array stored in the coefficient DB 224. The value is set to the coefficient K0 (corresponding to the first coefficient) of the digital filter 21 shown in FIG. 2 (step S04).

  Next, the second setting unit 225b uses the corrected distortion signal values at the positions A1 to A4 in the coefficient array stored in the coefficient DB 224 as the coefficients K1 to K5 (second values) of the digital filter 21 illustrated in FIG. (Corresponding to a coefficient) (step S05). Thereafter, the execution of the coefficient setting process is terminated. Note that the coefficients K1 to K5 of the tap on the output side of the 0th tap in FIG. 2 are voices a predetermined time before the signal that the digital filter 21 performs arithmetic processing using the coefficient K0 (corresponding to the 1st coefficient). This is a coefficient (corresponding to the second coefficient) used in the arithmetic processing for the signal.

  Here, the first setting unit 225a and the second setting unit 225b calculate an average value for each element of a plurality of coefficient arrays accumulated in the coefficient DB 224 (that is, a plurality of coefficient arrays calculated a plurality of times). And a coefficient array having the calculated average value as a coefficient element may be set in the coefficient array of the digital filter 21. According to this configuration, it is possible to reduce the influence of noise generated during measurement of the sound field on the coefficient value.

  The first setting unit 225a sets a reference signal value for the coefficient K5 of the digital filter 21, and the second setting unit 225a sets the sampling time from the coefficient K0 to K4 of the digital filter 21 in order of late sampling time. Each normalized corrected distortion signal value may be set. In this case, the coefficient K5 corresponds to the first coefficient, and the coefficients K0 to K4 correspond to the second coefficient. This is because the coefficients K0 to K4 are used in the calculation process for the audio signal after a predetermined time from the signal calculated using the coefficient K5.

Next, the impulse response of the digital filter 21 in which the coefficient array is set to be even symmetric will be described.
When an impulse signal is input, the digital filter 21 in which the coefficient array is set outputs an impulse response having phase distortion as shown in FIG. This phase distortion is observed as a distortion spectrum after the largest response observed at time 0 (that is, after the first sample). Among the responses, the signal shown in the fourth quadrant of FIG. 6A output by the digital filter 21 at the first sample is the distortion of the sound in the second sample shown in the first quadrant of FIG. This signal cancels out noise. That is, when the speaker 40 outputs the sound of the first sample shown in FIG. 5A to the sound field based on the signal output from the digital filter 21 at the time of the 0th sample, it starts to vibrate by outputting the sound. To do. Next, when the sound is output based on the signal output from the digital filter 21 at the time of the first sample, the speaker 40 performs an operation of canceling the vibration generated at the time of the 0th sample. For this reason, as shown in FIG. 6B, the distortion or noise in the sound field of the voice shown in the second sample in FIG. 5A is eliminated.

Similarly, the signal in FIG. 6A output from the second sample to the fifth sample by the digital filter 21 is a signal that cancels the distortion or noise of the sound in the third to sixth samples in FIG. 5A, respectively. is there.
For this reason, as shown in FIG. 6B, the impulse response result of the speaker 40 is the distortion shown in the first quadrant and the fourth quadrant in the impulse response result of the digital filter 21 shown in FIG. 6A. Alternatively, a response result in which noise is canceled is obtained. Therefore, the sound field correction apparatus 20 can output a sound that cancels the distortion or noise of the sound output from the speaker 40 based on the measurement result of the measurement sound in the sound field.

  That is, the response signal input from the microphone 50 matches the impulse signal input to the speaker 40. For this reason, the digital filter 21 can reduce the difference between the sound represented by the sound signal input to the digital filter 21 and the sound output from the speaker 40 based on the sound signal output from the digital filter 21. Here, since the impulse sound is a sound including all frequencies, the audio system 100 in which the coefficient is set by the coefficient setting device 22 does not change the position of the microphone 50 (that is, normal use) even if the frequency of the input signal is changed. It is possible to make the sound reach the user's listening position at a certain intensity with a certain strength. That is, the coefficient setting device 22 of the audio system 100 can set the coefficient array of the digital filter 21 according to the sound field so that the frequency-level characteristic of the digital filter 21 becomes flat.

  Further, according to the above configuration, since the sign of the measured distortion signal value is reversed and set to the coefficient array of the digital filter 21, the number of digital filters 21 that reduce distortion or noise of the sound collected by the microphone 50 is small. Can be generated with computational complexity.

  Furthermore, according to these configurations, for example, the coefficient array of the digital filter 21 that reduces the difference between the measurement signal output to the speaker 40 and the response signal input to the microphone 50 is calculated without using Fourier analysis and inverse Fourier analysis. it can.

  Furthermore, according to these configurations, for example, a coefficient array that reduces the difference can be calculated without using an LMS method (Least Mean Square). Therefore, not only can the amount of calculation for calculating the coefficient array be reduced, but also convergence can be achieved. There is no problem.

  Furthermore, since the audio system 100 can generate a coefficient array with a small amount of computation, instead of the coefficient setting device 22, representative function units (measurement signal output unit 221, sampling unit 222a, A configuration including a circuit that realizes a reference signal value selection unit 223a, a distortion signal value selection unit 223b, an inverse correction unit 223d, a normalization unit 223f, and a setting unit 225) can be employed. According to this configuration, for example, the coefficient array of the digital filter 21 can be set to have a flat frequency-level characteristic with a simple and inexpensive configuration that does not use a DSP (Digital Signal Processor) or a microcomputer.

  In the present embodiment, the sound field correcting device 20 outputs the measurement signal to the speaker 40 once, and based on the sampled response signal value of the microphone 50 that has collected the measurement sound (that is, sampling). A configuration in which the coefficient array of the digital filter 21 is set (without averaging the signal values) can be employed. In addition, the sound field correcting device 20 inputs the measurement signal to the digital filter 21 in which the coefficient array has already been set, and again, based on the response signal value of the microphone 50 that has collected the measurement sound, the coefficient of the digital filter 21. A configuration for setting an array can be employed.

  In the present embodiment, the audio system configured by the sound field correction device 20 may be built in the car navigation device, or may be used indoors or outdoors, for example, without being mounted on the vehicle. Furthermore, the sound field correction apparatus 20 may be incorporated in a CD player, a DVD player, a Blu-ray disc player, or a portable audio player.

In the present embodiment, the reference signal value selection unit 223a has been described as selecting the reference signal value based on the magnitude of the response signal value. However, the reference signal value selection unit 223a is based on the sampling time of the response signal value. The reference signal value may be selected. Specifically, the reference signal value selection unit 223a may first select a response signal value that exceeds a predetermined threshold as the reference signal value. This is because the distortion of the sound caused by the vibration of the speaker 40 often occurs after the sound output causing the vibration.
Similarly, the distortion signal value selection unit 223b may select a distortion signal value based on the sampling time of the response signal value and the sampling time of the reference signal value. Specifically, the distortion signal value selection unit 223b can adopt a configuration in which a response signal value sampled at a time later than the sampling time of the reference signal value is selected as a distortion signal value. Usually, the impulse response is because a plurality of distortion spectra having a signal value smaller than the signal value are generated after the largest signal value. According to these configurations, it is possible to accurately specify audio distortion caused by vibration of the speaker 40 caused by audio output.

  It should be noted that the present invention can be realized by a sound field correction device having the same functions and configurations as those of the sound field correction device 20 of the above-described embodiment, and also by applying a program to an existing sound field correction device. It is also possible to function as a sound field correcting device used for implementing the sound field correcting method using the digital filter and the coefficient setting method of the digital filter. In this case, a program for causing a computer (a control unit such as a CPU) having the same configuration as that of the sound field correction apparatus 20 exemplified in the above embodiment to realize the same function as the function of the sound field correction apparatus 20 described above. Can be made to function as a device used in the coefficient setting method according to the present invention.

  The application method of such a program is arbitrary, for example, it can be stored and distributed in a recording medium such as a memory card, CD-ROM, or DVD-ROM, and can also be distributed via a communication medium such as the Internet. .

DESCRIPTION OF SYMBOLS 10 Player apparatus 20 Sound field correction apparatus 21 Digital filter 22 Coefficient setting apparatus 30 Output circuit 40 Speaker 50 Microphone 22a CPU
22b RAM
22c ROM
22d Input unit 22e Output unit 221 Measurement signal output unit 222 Response signal acquisition unit 222a Sampling unit 222b Signal value storage unit 222c Signal value DB
222d Averaging unit 223a Reference signal value selection unit 223b Distortion signal value selection unit 223c Coefficient array generation unit 223d Reverse correction unit 223e Signal value storage unit 223f Normalization unit 223g Coefficient array storage unit 224 Coefficient DB
225 setting unit 225a first setting unit 225b second setting unit

Claims (7)

A sampling step of sampling the signal value of the signal output from the microphone that collected the measurement sound;
A first setting step of setting a first coefficient of a digital filter that outputs an audio signal to the speaker that outputs the measurement sound, to the signal value sampled in the sampling step;
A second coefficient used for arithmetic processing on the audio signal at a time different from the audio signal that the digital filter performs arithmetic processing using the first coefficient by a predetermined time is used as the first coefficient in the first setting step. A second setting step of setting to a reciprocal of the signal value sampled at a time different from the sampling time of the signal value set to the predetermined time;
A coefficient setting method for a digital filter, comprising: A reference signal value selection step of selecting a reference signal value to be set in the first coefficient from the sampled signal value based on the magnitude of the signal value sampled in the sampling step;
Based on the reference signal value selected in the reference signal value selection step and the sampled signal value, the distortion signal value representing the distortion of the sound output from the speaker is sampled. A distortion signal value selection step for selecting from the signal value obtained,
The first setting step sets the reference signal value selected in the reference signal value selection step to the first coefficient,
The second setting step sets a reciprocal of the distortion signal value selected in the distortion signal value selection step to the second coefficient;
The coefficient setting method for a digital filter according to claim 1. In the sampling step, when the speaker receives an impulse signal, the response signal value output from the microphone is sampled,
In the reference signal value selection step, the largest signal value among the response signal values sampled in the sampling step is selected as the reference signal value,
In the distortion signal value selection step, the signal value smaller than the reference signal value is selected as the distortion signal value among the response signal values sampled in the sampling step.
The coefficient setting method for a digital filter according to claim 2. A signal value storage step of storing in the storage device a plurality of correspondence information in which the response signal value sampled in the sampling step and the sampling time information indicating the sampling time of the response signal value are associated;
An average value calculating step of averaging the response signal value stored in the storage device in association with the sampling time information in the signal value storing step based on the sampling time information;
In the reference signal value selection step, the largest value among the response signal values averaged in the average value calculation step is selected as the reference signal value,
In the distortion signal value selection step, a value smaller than the reference signal value is selected as the distortion signal value among the response signal values averaged in the average value calculation step.
The coefficient setting method for a digital filter according to claim 3. Sampling means for sampling the signal value of the signal output from the microphone that collected the measurement sound;
First setting means for setting a first coefficient of a digital filter that outputs an audio signal to the speaker that outputs the measurement sound, to the signal value sampled by the sampling means;
A second coefficient used for arithmetic processing for the audio signal at a time different from the audio signal calculated by the digital filter using the first coefficient by a predetermined time is used as the first coefficient by the first setting means. Second setting means for setting to a reciprocal of the signal value sampled at a time different from the sampling time of the signal value set to the predetermined time;
A coefficient setting device for a digital filter, comprising: Computer
Sampling means for sampling the signal value of the signal output from the microphone collecting the measurement sound,
First setting means for setting, to the signal value sampled by the sampling means, a first coefficient of a digital filter that outputs an audio signal to a speaker that outputs the measurement sound;
A second coefficient used for arithmetic processing on the audio signal at a time different from the audio signal calculated by the digital filter using the first coefficient by a predetermined time is calculated by the first setting means by the first coefficient. Second setting means for setting a reciprocal of the signal value sampled at a time different from the sampling time of the signal value set by the predetermined time;
A coefficient setting program for a digital filter, which is made to function as: A sampling step of sampling the signal value of the signal output from the microphone that collected the measurement sound;
A first setting step of setting a first coefficient of a digital filter that outputs an audio signal to the speaker that outputs the measurement sound, to the signal value sampled in the sampling step;
A second coefficient used for arithmetic processing on the audio signal at a time different from the audio signal that the digital filter performs arithmetic processing using the first coefficient by a predetermined time is used as the first coefficient in the first setting step. A second setting step of setting to a reciprocal of the signal value sampled at a time different from the sampling time of the signal value set to the predetermined time;
The digital filter performs an arithmetic processing on the audio signal using the first coefficient whose value is set in the first setting step and the second coefficient whose value is set in the second setting step to the speaker. Output audio signal output step;
A sound field correction method using a digital filter characterized by comprising:

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