JPH05181489A - Sound field correction device - Google Patents

Abstract

PURPOSE:To automatically adjust the disorder of frequency characteristics owing to the influence of a sound field by finding the difference between frequency characteristics of a signal obtained by a listening means and frequency characteristics of a souce-side signal and correcting the frequency characteristics according to the difference. CONSTITUTION:The output of a preamplifier 1 is sent to a power amplifier 3 through a graphic equalizer 2 consisting of (n) electronic elements and a voice is generated by a speaker 4 and outputted. The graphic equalizer 2 initially has flat characteristics and an uncorrected voice signal is generated. For the purpose, a microphone 5 is installed at a specific listening point in a sound field formed by the speaker 4 and its signal is supplied to plural band- pass filters 7 through a microphone preamplifier 6. A signal which is divided into (n) audio bands by the band-pass filters 7 is passed through a sample holding circuit 8 and converted by an A/D converter 9 into a digital signal, and a microcomputer 10 controls the timing of sampling.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a sound field correction device.

[0002]

2. Description of the Related Art In audio equipment, sound emitted from a speaker is not affected by a sound field, such as reflection or diffraction, until reaching a listening point unless the room is a free sound field such as an anechoic room. It is known that, as a result, the frequency characteristic is disturbed due to the difference in frequency wavelength. A fixed equalizer, a graphic equalizer, a parametric equalizer, or the like is used to correct the disturbance of the frequency characteristic due to the influence of the sound field.

[0003]

However, the fixed equalizer described above has a drawback that it cannot respond to changes in the sound field at all. In the case of a graphic equalizer or a parametric equalizer, readjustment is possible, but there is a drawback that automatic adjustment cannot be performed because manual adjustment is necessary because it is necessary to rely on the sense of hearing or to set a measuring instrument. .. It is an object of the present invention to overcome the drawbacks of these conventional sound field correction devices.

[0004]

In order to achieve the above object, the sound field correction apparatus of the present invention comprises a listening means for listening to an acoustic signal at a predetermined listening point, converting it into an electrical signal and outputting the electrical signal. Means for obtaining the difference between the frequency characteristic of the source side signal output as an acoustic signal and the frequency characteristic of the heard signal for each predetermined frequency band, and performing the sound field correction based on the difference of the frequency characteristic And a sound field correction means.

[0005]

The difference between the frequency characteristic of the signal obtained by the listening means and the frequency characteristic of the signal on the source side is obtained, the frequency characteristic is corrected based on the difference, and the sound field correction is executed.

[0006]

Embodiments of the present invention will be described below with reference to the drawings. In FIG. 1, the output of the preamplifier 1 is sent to the power amplifier 3 via the graphic equalizer 2 of the electronic n element, and is output as voice from the speaker 4. The graphic equalizer 2 has a flat characteristic in the initial state, and an uncorrected audio signal is output from the speaker 4 via the power amplifier 3. A microphone 5 is installed at a predetermined listening point of a sound field formed by the speaker 4, and a signal from the microphone 5 is configured to be supplied to a plurality of bandpass filters 7 via a microphone preamplifier 6. The bandpass filter 7 divides the audible band into n predetermined bands, and the band is at least several bands, but the larger the number, the more accurate the correction becomes. The signal divided into n bands by the band pass filter 7 is sampled by the sample and hold circuit 8, converted into a digital signal by the A / D converter 9 and sent to the microcomputer 10. The timing of sampling by the sample and hold circuit 8 is controlled by the microcomputer 10.

On the other hand, the source signal from the preamplifier 1 is also sent to the bandpass filter 21 via the delay circuit 20, is divided into the above n frequency bands, is sampled by the sample and hold circuit 22, and is then A The / D converter 23 is configured to convert into a digital signal and input to the microcomputer 10. Delay circuit 2
0 is for delaying by the time required to reach the microphone 5 from the preamplifier 1, but a method of adjusting the control signal of the sample and hold circuit 22 is also possible.

The microcomputer 10 is a preamplifier 1
The source signal a input from the microphone and the feedback signal b input from the microphone 5 are processed as follows. As shown in FIG. 2, when a correction start operation command is issued by the operator, a sample signal is sent to the sample and hold circuit 8 and the sample and hold circuit 22,
Run the sample (step 30). When a predetermined time has elapsed, a hold signal is sent from the microcomputer 10 and the sampled analog signal is sent to the A / D converter 9
And digitized by the A / D converter 23 and read the data (step 31). After reading the data, the levels of the source signal a and the feedback signal b are adjusted (step 32). This corrects a difference in level between the source signal a and the feedback signal b due to the gain of the power amplifier, the efficiency of the speaker, the sensitivity of the microphone 5, the gain of the microphone preamplifier 6, and the like. And
The correction is executed (step 33) and the data is transferred (step 34).

The level matching described above will be described in detail.
As shown in FIG. 3, k is set to 0, and the difference between the source signal a and the feedback signal b is obtained from 1 to n for each frequency band (steps 40, 41, 42, 43). This is shown in FIG. Assuming that the difference between the source signal a and the feedback signal b is ck, the average value of this ck is obtained and the difference dk between ck and the average value in each frequency band is obtained (steps 44, 45, 46, 47). This d
FIG. 6 shows k and the source signal a.

This dk represents the difference between the source signal a and the feedback signal b after the level adjustment, and the sound field correction is performed based on this dk. That is, dk shown in FIG.
Is due to the influence of the sound field, and thus the dk can be corrected by equalizing the characteristic opposite to the frequency characteristic. Therefore, as shown in FIG. 4, first, dk is subtracted from the source signal a having a flat characteristic for each frequency band to obtain ek as shown in FIG. 7 (step 50).

However, it is not necessary to simply convert the electric signal into an acoustic signal, and since it is necessary to consider the auditory sensitivity, an auditory sense correction (step 51) is added. A method based on the Fletcher / Manson curve, the Robinson curve, or the like is known for the auditory sense correction, and any conventionally known method can be adopted. In this embodiment, a simplified method is used. As shown in FIG. 8, the audibility characteristic fk is subtracted from the eknes characteristic ek before the audibility correction for each frequency band to obtain the audibility correction characteristic gk.

The sensitivity of the microphone 5 is not necessarily flat in the audible band of 20 Hz to 20 KHz, and the low-end and high-end sensitivities are often poor due to the caliber of the microphone. Therefore, microphone correction (step 52) is further added. This correction is performed based on the sensitivity data of the microphone 5 used. FIG. 9 shows an example thereof. The microphone characteristic pk is subtracted from the uncorrected characteristic gk for each frequency band to obtain the corrected characteristic hk.

This data is used as the final correction characteristic zk as shown in FIG. 10, and the gain of each element of the graphic equalizer 2 is set (step 53). Then, this data is transferred from the microcomputer 10 to the graphic equalizer 2
, And the sound field is corrected by applying equalizing at the front stage of the power amplifier 3.

As described above, in the above embodiment, the microphone 5 feeds back the signal indicating the disturbance of the frequency characteristic due to the sound field, and the graphic equalizer 2 performs the equalizing so as to cancel the disturbance. The sound field is automatically and accurately corrected.

[0015]

As described above, the sound field correction apparatus of the present invention listens to an acoustic signal at a predetermined listening point, converts the electrical signal into an electrical signal, and outputs the electrical signal as an acoustic signal. A means for obtaining a difference between the frequency characteristic of the source side signal and the frequency characteristic of the listened signal for each predetermined frequency band, and a sound field correction means for executing a sound field correction based on the difference in the frequency characteristic. Therefore, the difference between the frequency characteristic of the signal obtained by the listening means and the frequency characteristic of the signal on the source side is obtained, the frequency characteristic is corrected based on the difference, and the effect of executing the sound field correction is obtained. is there.

[Brief description of drawings]

FIG. 1 is a block diagram showing an embodiment of the present invention.

FIG. 2 is a flow chart for explaining the operation of the embodiment of the present invention.

FIG. 3 is a flow chart for explaining the operation of the embodiment of the present invention.

FIG. 4 is a flow chart for explaining the operation of the embodiment of the present invention.

FIG. 5 is a waveform diagram illustrating the operation of the embodiment of the present invention.

FIG. 6 is a waveform diagram illustrating the operation of the embodiment of the present invention.

FIG. 7 is a waveform diagram illustrating the operation of the embodiment of the present invention.

FIG. 8 is a waveform diagram illustrating the operation of the embodiment of the present invention.

FIG. 9 is a waveform diagram illustrating the operation of the embodiment of the present invention.

FIG. 10 is a waveform diagram illustrating the operation of the embodiment of the present invention.

[Explanation of symbols]

1: Preamplifier, 2: Graphic equalizer, 3: Power amplifier, 4: Speaker, 5: Microphone, 6: Microphone preamplifier, 7: Bandpass filter, 8:
Sample and hold circuit, 9: A / D converter, 10: Microcomputer, 20: Delay circuit, 21: Bandpass filter, 22: Sample and hold circuit, 23: A
/ D converter.

Claims (1)

[Claims] 1. A listening unit that listens to an acoustic signal at a predetermined listening point, converts the acoustic signal into an electrical signal and outputs the electrical signal, and frequency characteristics of a source-side signal that is output as the acoustic signal.
A means for obtaining a difference from the frequency characteristics of the listened signal for each predetermined frequency band, and a sound field correction means for executing a sound field correction based on the difference in the frequency characteristics. Sound field correction device.