JPS6075199A - Electroacoustic transducer - Google Patents

Abstract

PURPOSE:To prevent deterioration of a characteristic of the 2nd harmonic distortion by leading a signal of an audible frequency band to an ultrasonic wave vibrator, converting the signal into a sound wave of a definite amplitude level, irradiating the wave in a medium such as air and reproducing the original audible sound through the nonlinear effect of the medium. CONSTITUTION:A signal generated from a modulation signal source 1 of an audible frequency band is fed to an adder 4 via coefficient multiplier 2 and added to a DC component from a DC source 3 at a prescribed rate at the adder 4. The added signal is given to a half power converter 5 and to a multiplier 7 after square root processing. The signal is multiplied with a carrier having a high frequency outputted from an ultrasonic wave band oscillator 6 at the multiplier 7, and the multiplied signal is given to an ultrasonic wave vibrator array 9 via a power amplifier 8. Thus, the sound wave of a definite amplitude level having a good 2nd harmonic distortion characteristic is generated from the ultrasonic wave vibrator array 9 and the wave is reproduced into an audible tone by the nonlinear effect of the medium.

Description

DETAILED DESCRIPTION OF THE INVENTION [W1 Public] The present invention relates to an electroacoustic transducer for radiating an electrical signal in an audible frequency band into the air as an acoustic signal.

Hongbai 3 Currently, the main types of electroacoustic transducers are electrodynamic direct radiation speakers and horn rotor speakers, but both methods create compression waves in the air by vibrating a diaphragm in the air. Sound engineer 2 for mechanical vibration energy
This is to convert it to a key.

The present invention utilizes a means completely different from conventional acoustic transducers such as speakers, that is, the parametric effect of finite amplitude sound waves due to air nonlinearity. Sound waves (
(referred to as the secondary wave).

It is characterized by a directivity pattern similar to that of carrier sound waves in the ultrasonic range.

Ultrasonic waves that have been amplitude-modulated by a signal in the IiT audio frequency band are radiated into a medium such as air or water at a finite amplitude level, and the self-demodulation effect that takes into account the nonlinear effects of the medium causes the ultrasonic waves to be modulated in the medium. Regarding the method of using the demodulated sound waves generated in the It has been reported. Parametric speakers that utilize this nonlinear phenomenon of sound waves are characterized by their sharp directivity, which is because they amplitude-modulate a high-frequency carrier wave with an audible signal wave and transmit it as a finite amplitude wave. Due to the nonlinear phase η action of the sound wave, the secondary waves related to the signal are divided into gate-shaped arrays in space! As a result of Ij, the directivity is sharp and the directivity becomes small.

A circular transmitter with radius a generates a finite amplitude sound wave P1= Po f (t) sin (+10 t) with envelope f (t).
=...=(A) is assumed to be radiated. Here, PO is for the source sound, and ω0 is the angular frequency of the carrier wave. If this 1
Assuming that the secondary wave or plane wave is sufficiently collimated, the secondary wave P2 will be along the sound axis. Note that β is a nonlinear parameter of the medium, ρ0 is the medium density, CO is the speed of sound, and α is the linear absorption coefficient of the primary wave. According to equation (2), P2 is proportional to f2. That is, it undergoes a nonlinear operation called the square of the envelope to produce a secondary wave. This square operation is a direct result of the second-order Ij latitude line of the sound wave, and is a cause of distortion in the reproduced signal. Therefore, in the current case, Pl is proportional to the reliability s (t), and no distortion occurs. Modify this modulation with two-sided waves t? (MDSB)
We will refer to this as a method.

Table 1 is based on the modulation method used in parametric loudspeakers, that is, the double sideband (DSB) method.
This, single sideband (SSB) method, and modified double sideband (MDS) method
The characteristics are compared with the E) method and expressed in dB.

Table 1 IJ, L, the signal is a sine wave, 100% modulated, and in equation (3), ! ■The case of T was targeted. As a result, DSB is inferior in terms of distortion or has the best conversion efficiency, while MDSB is superior in that it produces no distortion. However, for this purpose, by making the signal broadband, the distortion will increase unless a transmitter that transmits the signal in that band is used.

The experimental results of the efficiency and distortion of the DSB and MDSB methods, in which 581 ultrasonic transducers with a center frequency of 40 KHz were arranged in a plane, are shown in FIGS. 1 and 2, respectively. However, in the 1st +'3, the reproduction sound pressure distance is 9.5 m when the peak value of the applied pressure to the speaker is constant, and in Figure 2, the 2nd and 3rd with the fundamental wave in OdB. It affects the frequency characteristics of harmonics. In addition, for the efficiency comparison experiment, the amplitude 11mlted condition was used with the peak value of Ichikawa applied to the speaker being constant (
There are two types of 9r, and both methods achieve the same level of sound pressure, which is listed as a theory. As for the j-distortion shown in Figure 2, the MDSB method is D as far as the second harmonic is concerned.
It is approximately 1 OdB smaller than SB.

However, the third harmonic tends to become larger because the vibrator does not have a wide band.

Therefore, the -1-power distribution formula has the disadvantage that there are many second harmonic distortion components, and in particular, the second harmonic distortion rate is directly related to the modulation degree during amplitude modulation. The deeper the high pitch, the worse it gets.

Purpose The present invention solves the above-mentioned drawbacks by emitting ultrasonic waves that are amplitude-modulated by a frequency band signal into the air at a finite amplitude level, and generates an audible signal that is self-demodulated by the nonlinear effect of the air. This was done with the aim of preventing deterioration of second harmonic distortion characteristics, which is a drawback in speakers that obtain signals (referred to as parametric speakers).

Configuration The configuration of the present invention will be described below based on examples.

-6, as the frequency of the ultrasonic waves increases, the sound waves emitted from the vibrator become beam-shaped and travel straight.

Now, if we assume that amplitude-modulated ultrasonic waves are emitted in the form of a beam from a transducer array with radius a, then
The sound pressure P at a point of distance can be expressed by the following equation.

0 ・・・・・・・・・ (4) T, Co is the speed of sound, α is the ilJ of the sound wave at each frequency ω0
The amplitude coefficient, PO is the initial sound pressure, m is the modulation degree, and g (t) is the modulation wave. The sound pressure of a secondary wave generated by demodulating the finite amplitude level ultrasonic wave expressed by equation (3) by a nonlinear parametric waveform in the air is expressed by the following It homogeneous wave equation.

・・・・・・・・・(5) In Equation (5), Ps: sound pressure of secondary wave, pa/density of air, q: virtual sound source density of secondary wave occurring in the primary wave heam, However, q is expressed by the following formula.

Therefore, from equations (4) and (6), the distance from the array fliX(
By calculating the virtual sound source density at a point (on the axis), the following equation is obtained [-The first term on the right side of equation (7) is the virtual sound source density based on the signal component.
``teIg'' represents the density, and the second term represents the virtual sound source density of the blur component.

As explained below, the present invention is based on the nonlinear parameter 4.
The present invention relates to a modulation method for removing distortion components generated in an acoustic transducer that utilizes this effect. In other words, this is a modulation method in which a certain DC component is added to the modulation signal, subjected to J-conversion, and then the product with the carrier signal is calculated.

In this case, the modulated signal can be expressed by the following equation.

becomes. Therefore, the sound pressure of the primary wave (modulated ultrasonic wave) at a distance of X from the transducer array becomes loud. 2 in this case
The virtual sound source density of the next wave is obtained using equation (6). However, when this modulation method is used, the distortion component disappears, as shown in the second term on the right side of equation (7), and it can be expected that the quality of reproduced sound will be significantly improved.

[Example 1] An example of a basic configuration for carrying out the present invention is shown in FIG.

In Figure 3, ■ is a modulation signal source (audio frequency band), 2
is a coefficient unit, 3 is a DC source, 4 is an adder, 5 is a J-converter,
6 is an ultrasonic band oscillator, 7 is a multiplier, 8 is a power amplifier, and 9 is an ultrasonic transducer array.

[Embodiment 2] A modified embodiment of the present invention is shown in FIG. In the same figure, l
O is a double integrator, and the others are as in Figure 3. The reproduced sound pressure obtained in a nonlinear parametric speaker based on this modulation method is given by the following equation at a point X on the axis of the array.

In other words, the reproduced sound pressure is proportional to the second-order differential of the original modulation signal.

Therefore, as shown in FIG. 4, by passing the modulation signal through a double integrator before modulation and then applying modulation, the reproduced sound FJ is proportional to the original modulation signal, that is, (12 ) can be obtained.

As is clear from the above description, according to the present invention, harmonic distortion of reproduced sound is improved and high quality reproduced sound can be obtained. In the conventional method, when the modulation degree (m≦1) is found, the distortion rate deteriorates significantly, or in the present invention, the distortion can be basically reduced regardless of m, so when m is large (however, m≦
The effect on 1) is significant. Here, since the reproduced sound pressure is proportional to m, being able to use a large m is a very desirable direction for improving the efficiency of the acoustic transducer.

[Brief explanation of drawings]

FIG. 1 is an efficiency comparison diagram of an ultrasonic transducer, FIG. 2 is a distortion comparison diagram, and FIGS. 3 and 4 are electrical circuit diagrams for explaining the present invention in detail. DESCRIPTION OF SYMBOLS 1... Modulation signal source, 2... Coefficient unit, 3... DC source, 4... Adder, 5... 4-converter, 6... Ultrasonic band oscillator, 7... - Multiplier, 8... Power amplifier, 9... Ultrasonic transducer array, 10... Double integrator. Patent applicant: Ricoh Co., Ltd. Figure 1 Frequency (KHz) Figure 2 Frequency (KHz) Figure 3 Figure 4 (71)

Claims (2)

[Claims] (1) The signal from the signal source in the audible frequency band and the DC component are added at a certain rate, subjected to 172nd power conversion processing, and then multiplied by a carrier of a higher frequency above the signal frequency. After amplifying the force, it is guided to an ultrasonic transducer that performs electroacoustic conversion, and the modulated signal is converted into a sound wave with a finite amplitude level, which is radiated into a medium such as air, and the nonlinear effects of the medium are An electroacoustic transducer characterized in that it generates an original audible sound by p). (2) The electroacoustic conversion according to claim (1), characterized in that the signal of the OT @ frequency band j+ii is passed through two double integrator numbers in advance (integrated twice between 11P). Device.