JPH11340764A - Graphic equalizer - Google Patents

Abstract

PROBLEM TO BE SOLVED: To provide a graphic equalizer capable of adjusting a phase difference for each frequency band generated before reaching a listening point and reproducing music by further higher sound quality. SOLUTION: A delay time calculating/setting part 14 controls switches SW0 -SWn so as to successively input signals for delay time measurement outputted from a sound source 13 for delay time measurement to band-pass filters BP1 -BPn . Then, the time after inputting the signals from the sound source 13 for the delay time measurement until inputting sound detection signals from a microphone 16 is measured, delay time for each frequency band is calculated and the delay amount of delay devices TD1 -TDn is set based on the result.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

The present invention relates to a graphic equalizer for an audio system, and more particularly, to a graphic equalizer having a function of controlling a delay time for each frequency band.

[0002]

2. Description of the Related Art In recent years, graphic equalizers have been widely used in car audio systems. The graphic equalizer is a device that divides an audible frequency into a plurality of bands and can adjust a level for each frequency band. With the graphic equalizer, the frequency characteristics of the audio device can be corrected flat or adjusted to the characteristics according to the user's preference.

[0003]

A conventional graphic equalizer simply divides an audio signal into a plurality of frequency bands and enables the gain of each frequency band to be set arbitrarily. Changes are not taken into account. However, as a result of experiments by the inventors of the present application, it has been found that the time required to reach a listening point (listening position) differs for each frequency band even for a sound output from one speaker. That is, 1
Even if the sound is output from one speaker, a phase difference occurs depending on the frequency band before the sound arrives at the listening point, which causes deterioration in sound quality. The delay amount (phase difference) of each frequency band is unique to the speaker, and when the speaker is changed, the delay amount for each frequency band also changes.

Accordingly, an object of the present invention is to provide a graphic equalizer which can adjust a phase difference for each frequency band which occurs until a listening point is reached, and which can reproduce music with higher sound quality.

[0005]

An object of the present invention is to provide an audio signal input terminal, and a plurality of delay units for delaying and outputting an audio signal input through the audio signal input terminal by a set time. A plurality of band-pass filters provided corresponding to the respective delay units, a delay-time measuring sound source for outputting a delay-time measuring signal, and an input for switching the plurality of band-pass filters for the delay-time measuring. Switching means for connecting to an output of a sound source or an output of a corresponding delay device; an adder for adding signals output from the plurality of bandpass filters to supply the output to a speaker; and detecting a sound output from the speaker. A delay time setting means connected to the sound detection means for individually setting signal delay amounts of the plurality of delay devices. Solved by the equalizer.

In this case, the switching means sequentially connects the input of each band-pass filter to the output of the delay time measuring sound source, and the delay time setting means outputs the delay time measuring sound source from the delay time measuring sound source. After the signal is output,
It is preferable that a delay time is calculated by measuring a time until a sound is detected by the sound detecting means, and a signal delay amount of the delay device is set based on the result.

Hereinafter, the operation of the present invention will be described. In the graphic equalizer of the present invention, a sound source for measuring delay time is provided, and one of the plurality of band-pass filters is connected to the sound source for measuring delay time by the switching means. Then, the delay time measurement signal output from the delay time measurement sound source is input to the band-pass filter, and after passing only a signal in a predetermined frequency band, the signal is supplied to the speaker, and the speaker emits sound. The sound emitted from the speaker propagates in the air, reaches sound detection means arranged at the listening point, and is detected by the sound detection means. The delay time setting means measures the time from when the delay time measurement signal is output from the delay time measurement sound source to when the sound is detected by the sound detection means to calculate the delay time. The delay time is calculated for each bandpass filter as described above, and based on the result, the signal delay amount of each delay device is individually set.

Thus, the delay time is adjusted for each frequency band (pass frequency band of the band-pass filter),
When reaching the listening point, the phase of each frequency band becomes the same. Therefore, high-quality music reproduction without turbidity is possible. The delay time setting means calculates a difference with the other delay times based on the longest delay time among the delay times calculated for each frequency band passing through each bandpass filter, and calculates the difference value in each bandpass filter. This is the time for delaying the signal at the delay device corresponding to the filter. In this case, the delay device corresponding to the band-pass filter of the frequency band having the longest delay time has a signal delay amount of zero.

The delay time setting means uses white noise as the sound source for delay time measurement, and performs cross-correlation between the signal for delay time measurement input from the sound source for delay time measurement and the signal input from the sound detection means. It is preferable to calculate the delay time based on the time when the maximum value is obtained. The sound output from the speaker has a distorted waveform due to the propagation characteristics from the speaker to the sound detection means, but the point at which the sound wave having the highest cross-correlation value is considered to be the arrival time,
By detecting the maximum value of the cross-correlation, the delay time until the sound reaches the sound detecting means can be accurately calculated.

The delay time setting means uses white noise as a sound source for delay time measurement, and an error signal between a signal for delay time measurement input from the sound source for delay time measurement and a signal input from the sound detection means. The delay time may be calculated based on a filter coefficient of an adaptive filter that performs adaptive processing so that the power of the adaptive filter becomes minimum. Since the filter coefficient of the adaptive filter reproduces the impulse response detected by the sound detection means, by detecting the time at which the filter coefficient becomes maximum, the delay time until the sound reaches the sound detection means is reduced. It can be calculated accurately.

Further, the delay time setting means uses a time-stretched pulse as a sound source for delay time measurement, and convolves a signal input from the sound detection means with a signal obtained by inverting the time-stretched pulse on the time axis. The operation may be performed, and the time at which the result of the convolution operation is maximized may be detected. By performing such a convolution operation, it is possible to accurately calculate the delay time until the sound reaches the sound detection unit. The time-stretched pulse is a signal that is elongated on the time axis by changing the phase of the impulse in proportion to the square of the frequency, and has an advantage that it is not easily affected by sudden noise. .

[0012]

Embodiments of the present invention will be described below with reference to the accompanying drawings. (First Embodiment) FIG. 1 is a block diagram showing a graphic equalizer according to a first embodiment of the present invention.

The graphic equalizer of this embodiment includes an audio input terminal 11, a plurality of (n) sets of delay units TD1 to TDn, changeover switches SW1 to SWn, band pass filters BP1 to BPn, and amplifiers A1 to An.
And an adder 12 for adding the outputs of the amplifiers A1 to An.
And the delay time for each frequency band is calculated and each delay device TD
A delay time calculation / setting unit 14 for setting a delay time for each of 1 to TDn; a delay time measurement sound source 13 for outputting a delay time measurement signal; and a changeover switch SW0 for switching the output destination of the delay time measurement sound source 13. It consists of. Then, the speaker 15 is driven by the signal output from the adder 12. When measuring the delay time for each frequency band, the microphone 16 is arranged at the listening point, and the sound output from the speaker 15 is detected by the microphone 16 and input to the delay time calculation / setting unit 14.

An audio signal is input to the audio input terminal 11 from an audio device such as a tuner or a CD player. The audio signal input to the audio input terminal 11 is simultaneously input to each of the delay units TD1 to TDn. Each of the delay units TD1 to TDn delays and outputs a signal by the time set by the delay time calculation / setting unit 14.

The signals delayed by the delay units TD1 to TDn are input to bandpass filters BP1 to BPn via changeover switches SW1 to SWn. The changeover switches SW1 to SWn and the changeover switch SW0 are all controlled by the delay time calculation / setting unit 14. The bandpass filters BP1 to BPn have different pass frequency bands. For example, the band-pass filter BP1 passes only low-range signals, the band-pass filter BP2 passes only signals in a frequency band slightly higher than the band-pass filter BP1, and the band-pass filter BPn passes only high-range signals. And so on
The pass frequency bands of the band-pass filters BP1 to BPn are slightly shifted.

The amplifiers A1 to An amplify the signals of a predetermined frequency band which have passed through the preceding band-pass filters BP1 to BPn, respectively, and the adder 12 adds the signals of the respective frequency bands output from the amplifiers A1 to An. And output.
The amplification factors of the amplifiers A1 to An can be arbitrarily changed by the user. The signal output from the adder 12 is supplied to a speaker 15, and a sound (sound wave) is emitted from the speaker.

The delay time measuring sound source 13 outputs the delay time measuring signal when the delay time calculating / setting unit 14 sets the delay amount (signal delay time: the same applies hereinafter) of each of the delay units TD1 to TDn. Output. Sound source 1 for delay time measurement
3 is input to the delay time calculating / setting unit 14, and the changeover switch S
It is input to any one of the band-pass filters BP1 to BPn via W0 and the changeover switches SW1 to SWn. Further, the delay time for each frequency band is detected, and each delay unit T
When setting the delay amount of D1 to TDn, the microphone 16 is arranged at the listening position, and the microphone 16 and the delay time calculation / setting unit 14 are connected.

(Operation When Setting Delay Time) Hereinafter, in the equalizer of the present embodiment, each of the delay units TD1 to TDn
The operation when the delay time is automatically set will be described. First,
The delay time calculation / setting unit 14 controls the changeover switch SW0 and the changeover switches SW1 to SWn to electrically connect the output of the delay time measurement sound source 13 and the input of the bandpass filter BP1. When the delay time measurement signal is output from the delay time measurement sound source 13, only a predetermined frequency component of the delay time measurement signal is output to the band-pass filter B.
After passing through P1, it is amplified by the amplifier A1 and supplied to the speaker 15. Thereby, sound is output from the speaker 15 and propagates in the air to reach the microphone 16 arranged at the listening point. The microphone 16 converts the sound into an electric signal and outputs the sound signal to the delay time calculation / setting unit 14 as a sound detection signal.

The delay time calculation / setting unit 14 receives a delay time measurement signal from the delay time measurement sound source 13 and
The time until the sound detection signal is input from the microphone 16 is measured, and is defined as a delay time t1. Next, the delay time calculation / setting unit 14 switches the changeover switch SW0 to output the output of the delay time measurement sound source 13 and the band-pass filter BP.
Electrically connect to the 2 input. When the delay time measurement signal is output from the delay time calculation / setting unit 14, a predetermined frequency component of the delay time measurement signal passes through the band pass filter BP2, is amplified by the amplifier A2, and is supplied to the speaker 15. Is done. Thereby, the speaker 15
When the sound reaches the microphone 16, a sound detection signal is input to the delay time calculation / setting unit 14. The delay time calculation / setting unit 14 measures the time from the input of the delay time measurement signal from the delay time measurement sound source 13 to the input of the sound detection signal, and sets it as the delay time t2.

In this way, the delay time calculating / setting unit 1
Reference numeral 4 sequentially switches the changeover switch SW0 to measure the delay times t1 to tn for each signal passing through each band pass filter BP1 to BPn. Thereafter, the delay time calculating / setting unit 14 sets the delay amounts of the delay units TD1 to TDn based on the delay times t1 to tn. For example, assuming that the delay time t1 is the longest, the corresponding delay device TD1
And the delay amount of the delay unit TD2 is (t2 -t
1),..., The amount of delay of the delay unit TDn is set as (tn-t1).

After setting the delay amount of each of the delay units TD1 to TDn in this way, the delay time calculation / setting unit 14 switches the changeover switches SW1 to SWn to set the delay units TD1 to TDn.
TDn is electrically connected to the band-pass filters BP1 to BPn. Thereby, at the listening point, the phase difference for each frequency band is corrected, and high-quality music without turbidity can be heard.

From the viewpoint of measuring the delay time, it is preferable that the delay time detecting signal output from the delay time measuring sound source 13 is an instantaneous pulse signal (impulse). However, a speaker cannot be driven by an instantaneous pulse signal. Therefore, in the present embodiment, white noise or time-stretched pulse is used as the delay time detection signal.

Hereinafter, an example of the configuration of the delay time calculating / setting unit 14 when white noise or a time stretched pulse is used as the delay time detecting signal will be described. (Configuration Example 1 of Delay Time Calculation / Setting Unit) The delay time from when a signal is input to each of the band-pass filters BP1 to BPn until the sound wave reaches the listening point is determined by the input signal to each of the band-pass filters BP1 to BPn. By calculating the cross-correlation between the output signal of the micro-fan 46 and the output signal of the micro-fan 46, and calculating the maximum value time.

FIG. 2 is a diagram showing the configuration of the delay time calculating / setting unit 14 when calculating the delay time using the cross-correlation. The delay time calculation / setting unit 14 shown in FIG.
It comprises a cross-correlation calculation unit 21, a cross-correlation function maximum / delay time search unit 22, and a delay time setting unit 23. In addition, the delay time measurement sound source 15 combined with the delay time calculation / setting unit 14 outputs white noise as a delay time measurement signal.

The cross-correlation calculator 21 performs a cross-correlation calculation between the white noise output from the delay time measurement sound source 13 and the sound detection signal output from the microphone 16. This cross-correlation calculation is performed according to the following equation (1).

[0026]

(Equation 1)

Here, h indicates white noise input from the delay time measuring sound source 13 to the delay time calculating / setting unit 14, and g indicates a sound detection signal input from the microphone 16 to the delay time calculating / setting unit 14. , N represents the length of time taken into the cross-correlation calculation unit 21, and τ represents the sampling time interval. FIGS. 3A and 3B are schematic diagrams illustrating a method of detecting a delay time by a cross-correlation operation. As shown in FIG. 3A, a signal h input from the delay time measuring sound source 13 to the delay time calculating / setting unit 14 and a signal g input from the microphone 16 to the delay time calculating / setting unit 14 are used. There is a time delay of mτ between them. For example, as shown in FIG.
And the signal g is sampled, the value of g (n) is shifted by m (where m is 1 to N-1) with respect to h (n), and the product of h (n) and g (n + m) is calculated. The delay time mτ can be calculated by calculating the shift amount m that maximizes the value of the equation (1). In the example of FIG. 3B, m =
In the case of 2, the value of equation (1) takes the maximum value.

Maximum value of cross-correlation function / delay time search section 22
Is used to search for a value of m at which the cross-correlation R (mτ) becomes maximum, based on the result of performing the cross-correlation calculation by sequentially changing the value of m in equation (1) by the cross-correlation calculation unit 21. Is the delay time (t1, t2,..., Tn). The delay time setting unit 23 calculates the delay times t1, t2,..., T determined by the cross-correlation function maximum value / delay time search unit 22.
The maximum value is extracted from n, and the difference from the remaining delay time is calculated based on the maximum value. As described above, when the delay time t1 is the maximum, the remaining delay time is determined by the difference from t1. The delay time setting unit 23 sets the delay amount of each of the delay units TD1 to TDn based on the calculation result.

(Configuration Example 2 of Delay Time Calculation / Setting Unit) The microphone 16 detects the delay time from when a signal is input to each of the band-pass filters BP1 to BPn until the corresponding sound reaches the microphone 16. It can also be calculated by measuring the time when the impulse response is maximum.

FIG. 4 is a diagram showing a configuration of the delay time calculating / setting unit 14 when calculating a delay time using an impulse response. The delay time calculation / setting unit 1 shown in FIG.
Reference numeral 4 denotes an analog / digital (A / D) converter 31, a memory control unit 32, a memory 33, an averaging processing unit 34, a convolution operation unit 35, an impulse response maximum value / delay time search unit 36, and a delay time setting unit 37. It consists of. The delay time measuring sound source 13 combined with the delay time calculating / setting unit 14 outputs a time stretched pulse (time stretching pulse).

A time-stretched pulse is a signal whose frequency characteristic H (k) is represented by the following equation (2).

[0032]

(Equation 2)

Here, m is a coefficient indicating the degree of shifting the phase of each frequency within the time stretched pulse,
Any integer value. N is a coefficient that defines the time of generation of the time stretched pulse. Also, k is from 0 to N-
A is an integer up to 1, and a is determined by the third expression included in Expression (2) if m and N are determined. For example, when m = 0, a = 0, so that H (k) = e for all k.
xp (0) = 1, and each frequency component becomes an impulse concentrated without being dispersed.

The actual time-stretched pulse output from the delay time measuring sound source 13 is a signal obtained by performing an inverse Fourier transform on the above equation (2), and an example is shown in FIG. The time-stretched pulse shown in FIG.
= 256, and becomes a signal in which each frequency component is dispersed for a predetermined time according to the values of N and m. Therefore,
By setting the value of N to be large and the value of m to be large, the energy of each frequency component can be dispersed over a long period of time. The longer the generation time is, the longer the time required for measuring the delay time becomes.
And the value of m need to be set.

The A / D converter 31 is connected to the microphone 16
Performs sampling and quantization at a predetermined time interval on the sound detection signal output from, and outputs data of a predetermined number of bits. The memory control unit 32 controls the A / D at predetermined time intervals.
The data output from the converter 31 is sequentially stored in the memory 33. When the time stretched pulse is output once, an analog signal waveform output from the microphone 16 as a response to the time stretched pulse is converted into digital waveform data (hereinafter, referred to as A / D converter 31).
(Referred to as “time-stretched pulse response data”) and stored in a predetermined area of the memory 33. Memory 3
No. 3 has L storage areas as described above.
When L time-stretched pulses are repeatedly output from the delay time measurement sound source 13, the corresponding time-stretched pulse response data is stored in the L storage areas described above.

The averaging section 34 averages L time-stretched pulse response data stored in the memory 33. Assuming that the i-th response data is qi (n), the averaged response data q (n) is
It is calculated by the formula.

[0037]

(Equation 3)

The averaging section 34 averages L time-stretched pulse response data according to the equation (3). As a result, response data from which the influence of sudden noise has been removed can be obtained. Convolution operation unit 35
Is data p (-n) obtained by inverting the time-stretched pulse p (n) on the time axis to the averaged time-stretched pulse response data q (n) calculated by the equation (3).
Is convolved. FIG. 6 is a diagram showing a signal obtained by inverting the time stretched pulse on the time axis, and shows a waveform obtained by inverting the time stretched pulse corresponding to N = 256 shown in FIG. The data p (-n) actually used in the convolution operation unit 35 is obtained by converting the signal waveform shown in FIG. 6 into digital waveform data, and the sampling interval is the same as the sampling interval in the A / D converter 31. It is the same as the conversion interval.

The convolution operation in the convolution operation section 35 is performed based on the following equation (4).

[0040]

(Equation 4)

According to the equation (4), an impulse response is obtained by convolving the time-stretched pulse response signal with a signal obtained by inverting the original time-stretched pulse on the time axis. The impulse response maximum value / delay time search unit 36, to which the operation result of the convolution operation unit 35 is input, searches for the operation result that maximizes the impulse response, thereby obtaining the delay times t1, t2, and t2 in each frequency band. .., Tn are calculated.

.., Tn calculated by the maximum impulse response value / delay time search unit 80. The delay units TD1 to TD
Set each delay amount of n. (Configuration Example 3 of Delay Time Calculation / Setting Unit) As described above,
The delay time from when a signal is input to each of the band-pass filters BP1 to BPn until the corresponding sound wave reaches the microphone 16 is calculated by measuring the time at which the impulse response detected by the microphone 16 becomes maximum. However, an adaptive filter can be used to measure the time at which the impulse response is maximized.

FIG. 7 shows a delay time calculation when an impulse response is obtained by an adaptive filter to calculate a delay time.
FIG. 3 is a diagram illustrating a configuration of a setting unit. The delay time calculation setting unit 14 shown in FIG.
ast Mean Square) algorithm processing unit 42, adder 4
3, a filter coefficient maximum value / delay time search unit 44 and a delay time setting unit 45. The delay time measuring sound source 13 combined with the delay time calculating / setting unit 14 outputs white noise as a delay time measuring signal.

The adaptive filter 41 has a FIR (Finite Imp
ulse response) type digital filter configuration, and uses the tap coefficient vector (filter coefficient) W set by the LMS algorithm processing unit 42 to remove white noise input from the delay time measurement sound source 13. A predetermined FIR filter process is performed. By the way, LM
The S-algorithm processing unit 42 controls the filter coefficient W of the adaptive filter 41 so that the power of the error signal e obtained by subtracting the output signal of the adaptive filter 41 from the sound detection signal by the adder 43 is minimized. Therefore, the output signal of the microphone 16 and the output signal of the adaptive filter 41 are substantially the same, and the filter coefficient W of the adaptive filter 41 has substantially the same characteristics as the impulse response detected by the microphone 16.

Filter coefficient maximum value / delay time search section 44
Is the time when the maximum value of the impulse response is obtained, that is, the band-pass filter B is searched for the time when the maximum value is obtained among the elements of the filter coefficient W of the adaptive filter 41.
Times t1, t2, t1, t2, and t3 from when a signal is input to P1 to BPn until the corresponding sound wave reaches microphone 16.
.., Tn are calculated.

The delay time setting unit 45 calculates the delay time t calculated by the filter coefficient maximum value / delay time search unit 44.
, T2,..., Tn, the delay units TD1 to TDn
Set each delay amount. (Second Embodiment) FIG. 8 is a block diagram showing a graphic equalizer according to a second embodiment of the present invention.
8, the same components as those in FIG. 1 are denoted by the same reference numerals, and detailed description thereof will be omitted.

In this embodiment, unlike the first embodiment, no amplifier is provided between the band-pass filters BP1 to BPn and the adder 12. Further, an equalizer section 17 is provided downstream of the adder 12. The equalizer section 17 is configured by serially connecting a plurality of filters having peaks or dips in a part of the frequency band of the all-band pass filter. in this case,
In addition to the characteristics of the band-pass filters BP1 to BPn, the frequency characteristics can be adjusted by each filter in the equalizer unit 17.

In this embodiment, similarly to the first embodiment, when setting the delay amounts of the delay units TD1 to TDn, the changeover switches SW0 and SW1 to SWn are controlled to control the sound source 13 for delay time measurement. Are sequentially input to the band-pass filters BP1 to BPn, and the delay time is measured for each pass frequency band of each of the band-pass filters BP1 to BPn. Then, based on the result, the delay amount of each of the delay units TD1 to TDn is set. Also in the present embodiment, the same effect as in the first embodiment can be obtained.

In the above-described first and second embodiments, the case has been described where the delay amounts of the delay units TD1 to TDn are automatically set by the delay time calculation / setting unit 14. The delay amounts of TD1 to TDn may be manually adjusted.

[0050]

As described above, according to the graphic equalizer of the present invention, the delay time measuring signal output from the delay time measuring sound source is sequentially input to a plurality of band-pass filters and output from the speaker. The sound is detected by the sound detection means, the delay time for each frequency band is measured by the delay time setting means, and the signal delay amount of each delay device is set according to the result. Thereby, different delay times are corrected for each frequency band, and when the sound output from the speaker reaches the listening point, the phase of each frequency band becomes the same, and high-quality sound without muddiness is reproduced.

[Brief description of the drawings]

FIG. 1 is a block diagram showing a graphic equalizer according to a first embodiment of the present invention.

FIG. 2 is a diagram illustrating a configuration of a delay time calculation / setting unit when calculating a delay time using a cross-correlation.

FIG. 3 is a schematic diagram showing a method for detecting a delay time by a cross-correlation calculation.

FIG. 4 is a diagram illustrating a configuration of a delay time calculation / setting unit when calculating a delay time using an impulse response.

FIG. 5 is a diagram illustrating an example of a time stretched pulse.

FIG. 6 is a diagram showing a signal obtained by inverting a time stretched pulse on a time axis.

FIG. 7 is a diagram illustrating a configuration of a delay time calculation / setting unit when calculating a delay time by obtaining an impulse response using an adaptive filter.

FIG. 8 is a block diagram showing a graphic equalizer according to a second embodiment of the present invention.

[Explanation of symbols]

 Reference Signs List 11 input terminal, 12 adder, 13 sound source for delay time measurement, 14 delay time calculation / setting unit, 15 speaker, 16 microphone, 17 equalizer unit, TD1 to TDn delay unit, BP1 to BPn bandpass filter, A1 to An amplifier .

Claims (6)

[Claims] 1. An audio signal input terminal, a plurality of delay units for delaying an audio signal input via the audio signal input terminal by a set time and outputting the same, respectively, A plurality of bandpass filters provided, a delay time measurement sound source for outputting a delay time measurement signal, and an output of the delay time measurement sound source or a corresponding delay device by switching inputs of the plurality of bandpass filters. A switching unit connected to an output, an adder for adding signals output from the plurality of bandpass filters and supplying the added signals to a speaker, and a sound detection unit configured to detect sound output from the speaker, and And a delay time setting means for individually setting a signal delay amount of the delay device. 2. The switching means sequentially connects an input of each band-pass filter to an output of the delay time measuring sound source, and the delay time setting means outputs the delay time measuring signal from the delay time measuring sound source. The delay time is calculated by measuring the time from when is output until the sound is detected by the sound detection means, and the signal delay amount of the delay unit is set based on the result. Item 2. The graphic equalizer according to Item 1. 3. The delay time setting means calculates the delay time for each frequency band passing through each band-pass filter, and calculates a difference from other delay times based on the longest one of the delay times. The graphic equalizer according to claim 2, wherein the calculated difference value is set as a signal delay amount of a delay unit corresponding to each band pass filter. 4. The delay time measuring sound source outputs white noise as the delay time measuring signal, and the delay time setting means inputs a signal input from the delay time measuring sound source and an input from the sound detecting means. 3. The graphic equalizer according to claim 2, wherein a time at which a cross-correlation with a signal to be performed is maximized is calculated, and the delay time is obtained for each frequency band passing through each band-pass filter. 5. The delay time measuring sound source outputs white noise as the delay time measuring signal, and the delay time setting means inputs a signal input from the delay time measuring sound source and an input from the sound detecting means. An adaptive filter that performs adaptive processing so that the power of an error signal with respect to the signal to be processed is minimized, and obtains the delay time for each frequency band passing through each band-pass filter based on the filter coefficient of the adaptive filter. The graphic equalizer according to claim 2, wherein: 6. The delay time measuring sound source outputs a time stretched pulse as the delay time measuring signal, and the delay time setting means outputs the time stretched pulse to a signal input from the sound detecting means. Convolving a signal obtained by inverting the signal on the time axis, and calculating a time when the convolution operation result is maximum, thereby obtaining the delay time for each frequency band passing through each band-pass filter. Item 3. The graphic equalizer according to Item 2.