JPH01238298A - Frequency characteristic correcting device for speaker - Google Patents

Abstract

PURPOSE:To improve the instantaneousness and accuracy of a signal processing without using any special signal for control by connecting a measuring system and a reproducing system in a loop, and using an adaptive filter. CONSTITUTION:A microphone 14 is disposed opposingly in the front of a speaker 13 and they are connected in an acoustic space. Between the microphone 14 and a sound source signal input terminal part 10, a signal delay circuit 15, a characteristic setting part 16, and an error detecting part 17 are connected in this order toward the microphone 14. The delay time of the circuit 15 is set at a time generated by adding the half of the impulse response time of the adaptive digital filter part 11 to the acoustic wave propagation time between the speaker 13 and the microphone 14. The filter part 11 is self-adjusted so that the mean square of the error information signal values of errors between signals from the microphone 14 and output signals obtained from the detection part 17 becomes minimum. By setting the transfer function of the setting part 16 in one, the characteristic of the filter part 11 becomes inverse to that of the speaker 13, and the overall characteristic becomes flat.

Description

DETAILED DESCRIPTION OF THE INVENTION (Field of Industrial Application) The present invention relates to a device for correcting the frequency characteristics of a speaker.

(Prior art) Conventionally, this type of device first measures the frequency versus amplitude characteristics of a speaker using a calibration signal such as white noise as a sound source signal using a microphone placed oppositely in front of the speaker. Then, the inverse characteristic is calculated from this measurement result, and the parameters (coefficients) necessary for setting in the digital filter are derived by performing calculation processing such as inverse Fourier transform. It has been known to attempt to flatten the frequency characteristics of the speaker by inserting and connecting it between the sound source signal and the speaker and controlling it with a central processing unit (CPtJ).

(Problem to be Solved by the Invention) However, this conventional device requires a measurement system for calculating parameters to be set in the digital filter, and a sound reproduction system for processing the i'FA signal with the digital filter and amplifying and amplifying the sound. However, each of these systems is independent, and there is a problem in that signal processing between the measurement system and the reproduction system cannot be performed in real time. In addition, as a regulation signal used in the measurement system,
A special signal such as white noise that has a flat frequency vs. amplitude characteristic is required, and if this is not used, there is a problem in that it is not possible to derive coefficients to flatten the characteristic at the computer simulation stage. .

This invention was made in order to solve the above-mentioned problems, and its purpose is to loop-couple the measurement system and the reproduction system and use an adaptive filter. It is an object of the present invention to provide a frequency characteristic correction device for a speaker that can perform instant signal processing, improve accuracy, and automatically correct characteristics without using a standard signal for correction.

(Means for Solving the Problems) The gist of the present invention for achieving the above-mentioned object is to provide a speaker, a microphone disposed in front of the speaker, a sound source signal input terminal section, and a sound source signal input terminal section. an adaptive filter section connected between the speaker and the speaker, and a signal delay circuit section connected to the sound source signal input terminal section;
an error detection section that extracts an error between the output signal of the signal delay circuit section and the output signal of the microphone and supplies the error information signal to the adaptive filter section; the error detection section and the signal delay circuit section; and a characteristic setting section connected between the two and used to set desired characteristics in advance.
The signal delay circuit section has a signal delay time that is the sum of the sound wave propagation time between the speaker and the microphone plus a half time length of the impulse response of the adaptive filter section, and the adaptive filter section The present invention resides in a speaker frequency characteristic correction device characterized in that it squares the error information signal value supplied from the error detection section and self-adjusts it so that the average value thereof is minimized.

(Function) With this configuration, if a desired characteristic is set in the characteristic setting section, the adaptive filter section performs the adaptation 1'1lI111 based on the error information signal detected by the error detection section. It will be done. When this adaptive control is completed, the characteristics of the adaptive filter section become the inverse characteristics of the frequency characteristics of the speaker, taking into account the characteristics set in the characteristic setting section. Then, the characteristics of the speaker including the adaptive filter section become equal to the characteristics set in the characteristic setting section.

For example, if the characteristic setting section is set so that the frequency characteristic is flat, the frequency characteristic from the sound source signal input terminal section to the speaker output signal will be flat.

At this time, adaptive control in real time is achieved by loop-coupled control of the calculation measurement system for the parameters set in the adaptive filter and the sound reproduction system for the sound source signal.

In addition, by detecting the error between the sound source signal and the speaker output signal and performing adaptive control using an adaptive filter, adaptive signal processing control can be performed using a normal audio signal or music signal as a regulation signal. will be carried out.

In addition, by applying adaptive signal processing control, the frequency characteristics can be easily automatically equalized and corrected.

(Example) Next, one embodiment of this water light will be described based on the drawings.

FIG. 1 is a block diagram of a speaker frequency characteristic correction device. In FIG. 1, a sound source signal input terminal section 10 to which a sound source signal is applied, an adaptive digital filter section 11,
A speaker driving power amplifier 12 and a speaker 13 are connected in series.

A microphone 14 is disposed in front of the speaker 13 to face it on the front axis, and is coupled to the speaker 13 in an acoustic space. Between the microphone 14 and the sound source signal input terminal section 10, there is a signal delay circuit section 15, a characteristic setting section 16 for setting and adjusting a transfer function to set a desired frequency characteristic, and a connection between the characteristic setting section 16 and the microphone 14. An error in the output signal is detected and the error information signal is sent to the adaptive digital filter section 11.
The error detecting units 17 that supply the signals to each other are connected in turn.

The delay time of the signal delay circuit section 15 is set to a time equal to the sound wave propagation time between the speaker 13 and the microphone 14 plus one-half time length of the impulse response of the adaptive digital filter section 11.

The adaptive digital filter section 11 self-adjusts the amplitude/phase vs. frequency characteristics of the filter so that the average value obtained by squaring the error information signal value obtained from the error detection section 17 is minimized. It is configured to achieve the multiplicative error method.

If the transfer function of the characteristic setting section 16 is set to 1 (frequency domain), the characteristic of the adaptive digital filter section 11 becomes an inverse characteristic of the characteristic of the speaker 13, and becomes a flat characteristic as a whole.

FIG. 2 is a system block diagram showing another embodiment of the invention. In FIG. 2, it has a finite impulse response type filter section (FIR filter section) 18 and an adaptive digital filter section 19, and further includes a first filter section that interlocks with each other.
This embodiment differs from the first embodiment in that it includes a changeover switch 20 and a second changeover switch 21.

The coefficients are then copied from the adaptive digital filter section 19 to the finite impulse response filter section 18.

When measuring the coefficients to be set in the digital filter sections 18 and 19, the first and second changeover switches 20 and 21
For example, for several seconds to 2 to 3 seconds with the switch switched to the contact a side.
Perform adaptive movements for minutes. When the adaptive operation is completed, the changeover switches 20 and 21 are switched to the contact side, and the highlighted coefficients are transferred to the adaptive digital filter section 19.
The F[R filter section 18 has a master-slave configuration in which the sound source signal is reproduced in this state.

FIG. 3 shows a block diagram of one adaptive digital filter unit 11 including the error detection unit 17.

The signal at the contact a of the changeover switches 20 and 21 is input as the main human input, and the error detection unit 1 is input as the reference input.
The signal from the signal delay circuit section 15 of No. 7 is inputted and outputted as an error information signal.

These input/output signals are processed by an analog/digital signal converter (A/D), which has a 16-bit linear conversion function.
)22. It is connected and supplied to an 8-bit data bus line 25 via a digital-to-analog signal converter 23 and an upper and lower 8-bit selector 24.

This bus line 25 includes a plurality of digital signal processor sections 26 each having a 16x24 bit multiplier that performs signal processing of the input signal and the error information signal. /RA
A built-in 8-bit microprocessor unit section 27 of M is connected.

A programmable parameter setting switch section 28 is connected to this processor unit section 27, and it sets the parameters, sampling frequency, filter taps, etc. necessary for performing the adaptive signal processing using the above-mentioned minimum mean square error method. The number, gain constant, and other initial setting parameters are set.

Note that it has a synchronizing signal generating section 29 that receives the Zumbling frequency data from the parameter setting switch section 28 and supplies a synchronizing signal to each section.

FIGS. 6 to 9 show examples of characteristics obtained as a result of experiments using the apparatus shown in FIG. At this time, a woofer with a diameter of 38 cm is used as a speaker, the sampling frequency is 8 kHz, the frequency @ band is 3.4 kHz, and the adaptive digital filter sections 18 and 19 are 2
56 taps, and the transfer function of the characteristic setting section 16 was set to 1'' to try to flatten the characteristics.

FIG. 4 shows the amplitude/phase frequency characteristics of the speaker 13 measured at a position 1 meter away from the speaker 13 before performing adaptive control. In the figure, the solid line is the amplitude characteristic, and the dotted line is the phase characteristic (the same applies below).

The characteristics after sufficient time response control is performed using white noise as the sound source signal input are not shown in FIG.

Generally speaking, is the speaker 13 a multi-mode speaker? ! It is a coarse vibrating body, and it is theoretically difficult to achieve flat characteristics over a wide band.

However, when comparing the characteristics in Figures 4 and 5, it is clear that the frequency characteristics of the linear system with complex undulations in Figure 4 are different from 100 Hz to 100 Hz as shown in the amplitude phase characteristics in Figure 5. from 3
It can be seen that flattening is achieved with high precision down to around kilohertz.

Here, there is a characteristic slope of about 5 decibels around 3.4 kHz, but this is due to the influence of the analog low-pass filters located before and after A/D and D/Δ, respectively. That is, in this method, by removing the influence of this analog low-pass filter, the corrected frequency characteristics with higher accuracy than those shown in FIGS. 4 and 5 can be improved by 1q.

Next, FIGS. 6 and 7 show the frequency characteristics before and after adaptation when the microphone 14 is positioned at the aperture of the speaker 13, assuming practical use.

The effects of frequency flattening will continue to be confirmed.

Furthermore, as an example of using something other than white or noise as the sound source signal, Figure 8 shows the results of adaptive operation with rock music with vocals (3 minutes and 30 seconds), and Figure 9 shows the long-term spectrum of the music signal. are shown respectively.

Comparing the characteristics in FIG. 7 and FIG. 8, it can be seen that the effects of flattening the amplitude/phase frequency characteristics are almost the same, and that the adaptive operation functions effectively even for music signals.

In addition, experiments were conducted with male announcement signals and classical music signals, and the same effect of flattening was confirmed.

Note that desired characteristics other than flat characteristics can be obtained by setting them in advance in the characteristic setting section 16.

(Effect of the invention) As mentioned above, according to this invention, frequency (horizontal axis) versus amplitude
It is possible to automatically equalize and correct the phase (vertical axis) characteristics at the time of rubbing to achieve the desired characteristics, and it is also possible to use natural sound source signals (voice, music), and to perform automatic correction of characteristics with high precision. The remarkable effects that can be achieved are obtained.

[Brief explanation of the drawing]

FIG. 1 is a block diagram showing an embodiment of the present invention, FIG. 2 is a block diagram showing an embodiment of the invention, FIG. 3 is a block diagram of an adaptive filter section including an error detection section, and FIGS. 9th
The figure is a frequency characteristic diagram showing an example of experimental results. 10... Sound source signal input terminal section, 11, 19... Adaptive filter section, 13... Speaker, 14... Microphone, 15... Signal delay circuit section, 16... Characteristic setting section, 17...Error detection section, 18...Finite impulse response type (FIR) filter section, 20, 21... Changeover switch section. Figure 1 Figure 1

Claims (1)

[Claims] (1) a speaker, a microphone placed in front of the speaker, a sound source signal input terminal section, an adaptive filter section connected between the sound source signal input terminal section and the speaker, and the sound source signal input terminal section; a signal delay circuit section connected to the terminal section; an error detection section that extracts an error between the output signal of the signal delay circuit section and the output signal of the microphone and supplies the error information signal to the adaptive filter section; ,
A characteristic setting section is provided which is connected between the error detection section and the signal delay circuit section and sets desired characteristics in advance, and the delay circuit section adjusts the sound wave propagation time between the speaker and the microphone. The signal delay time is the sum of half the time length of the impulse response of the adaptive filter section, and the adaptive filter section squares the error information signal value supplied from the error detection section. A frequency characteristic correction device for speakers, which is characterized by self-adjusting so that the average value thereof is minimized.