KR101882140B1 - Complex speaker system capable of ultra directional and non directional simultaneous signal output - Google Patents

Abstract

The present invention relates to a complex speaker system capable of simultaneous output of super-directional and omni-directional signals. The complex speaker system includes: a signal input unit for inputting an audio input signal to be amplified; an A/D converter for converting an analog input signal into a digital signal; a signal delay unit for delaying and outputting the omni-directional signal by a delay time of a signal output from a super-directional speaker system; a pulse width modulator for pulse-width-modulating the signal output from the signal delay unit; and an omni-directional amplifier which amplifies and outputs the pulse-width-modulated signal.

Description

BACKGROUND OF THE INVENTION 1. Field of the Invention [0001] The present invention relates to a composite speaker system capable of simultaneously outputting a super-directional signal and an omnidirectional signal,

The present invention relates to a composite speaker system capable of simultaneously outputting a supergain signal and an omnidirectional signal, and more particularly, to an omnidirectional signal output for an input audio signal at the same time as a supergain signal output, Directional and omnidirectional simultaneous signal output that enables the signal to be output simultaneously, and a new signal processing method that can improve the sound quality of the ultrasonic speaker system. ≪ / RTI >

Generally speaking, a speaker generates sound by converting an electric signal into vibration and delivering it to the air. These loudspeakers are isotropic and transmit when transmitting vibrations in the air. Accordingly, the listener can hear the sound generated from the speaker in all directions with respect to the speaker. The isotropy of these loudspeakers often causes unnecessary problems. For example, if a variety of works or exhibits such as a museum or a museum are installed and the installation is designed to explain the description of each work and the exhibits through a speaker, Interference occurs between sounds. Also, if many people hear the descriptions of different works and exhibits at the same time from the speakers, a large amount of voice guidance is interfered with and distorted from each other, and becomes a huge noise. In order to solve this problem, a super directional speaker which can reproduce a sound in a specific direction has emerged.

However, it is pointed out that the supergain speaker can not simultaneously transmit information to listeners scattered in various directions. For example, it may be necessary to deliver an omnidirectional voice in order to communicate an urgent situation or to convey specific information to everyone in the vicinity.

For this purpose, a speaker system in which a supergiant speaker and a non-directional speaker are mixed is required. However, in general, a signal outputted from a supergain speaker is delayed due to a predetermined signal processing, There is a problem that information can not be delivered to the listener properly.

In addition, one of conventional conventional directional speaker methods uses a parabolic dish, which is a speaker designed to have a general speaker installed at the focus of a parabolic dish so that the acoustic output of the speaker is reflected on the dish and straightened. This parabolic super-directional speaker is often known as a museum speaker because it is often used in museums. However, in the conventional supergain speaker system using the parabolic dish, the diameter of the parabolic dish is considerably large, and the distance to which the sound is transmitted with a direction is short as short as 10 m, and sound quality is also considerably deteriorated.

Accordingly, recently, ultrasonic speaker technology using nonlinear interference phenomenon caused by ultrasonic waves in air has been widely applied to realize a supergain speaker. Ultrasonic speaker technology has been developed since the 1960s, but commercialization has been delayed due to poor development of various peripheral devices and industrial gain problems, and it is being developed in earnest recently.

The supergain speaker system is composed of a signal processing unit for obtaining an appropriate sound quality, a modulating unit for efficiently modulating the processed signal in the ultrasonic band, an ultrasonic wave amplifying unit for driving the ultrasonic wave changer, and an ultrasonic transducer for generating ultrasonic waves in the air . Theoretically, the audible signal p (t) demodulated in air is proportional to the second time derivative of the square of the envelope E (t) of the amplitude modulated signal as shown in equation (1). In Equation (1), the second time differential can be solved by using an equalizer of 12 dB / octave, and the envelope E (t) can be expressed as Equation (2).

[Equation 1]

^{p (t) α ð 2 /} ðt 2 {E (t) 2}

&Quot; (2) "

E (t) = 1 + mx (t)

Where m is the modulation index and x (t) is the original audible audio signal.

In the above equation, the audible signal p (t) heard through the ultrasonic speaker becomes audible sound reproduction free from distortion when proportional to the original audible audio signal x (t). In practice, however, the distortion corresponding to the square of the original signal x (t) occurs as in Equation (1). As a method of reducing the distortion, a conventional ultrasonic speaker can reduce the distortion by reducing the modulation index m, but it is difficult to obtain a high sound output because the reproduction efficiency is low.

The distortion compensation method for compensating for the other distortion is a method of modulating the square root of the original signal as shown in FIG. Theoretically, according to this method, the original signal can be reproduced faithfully, but the spectrum of the original signal x (t), whose bandwidth is limited by the nonlinear calculation of square root, appears on almost infinite bandwidth. Thus, without an ultrasound transducer to reproduce an infinite bandwidth, the ultrasonic speaker of the type shown in FIG. 1 has absolutely limitations in reducing distortion.

In order to solve the problem as shown in FIG. 1, ATC of the United States has developed a repetitive error compensation method without increasing the bandwidth in the patent application entitled " Modulator Processing for a Parametric Speaker System (US 6,584,205) " . In brief, the patents invented by ATC in the US are calculated by calculating the ideal modulated signal waveform through a SSB (Single Side Band) channel model without a converter, then calculating the error by comparing it with the actual modulated signal, And compensates the distortion of the sound quality by repeating the process of compensating the signal several times. However, ATC's invention patents repeatedly compensate for errors. In order to compensate for such errors repeatedly, a large amount of computation is required. Therefore, the hardware design becomes complicated and the latency due to signal processing increases. There are disadvantages. In addition, the invention patent of ATC Company of the United States adopts SSB modulation. In order to avoid distortion due to incompleteness of the SSB filter, it is necessary to design a sharp SSB filter by increasing the degree of the SSB filter.

SUMMARY OF THE INVENTION It is an object of the present invention to solve the above-mentioned problems, and it is an object of the present invention to provide an apparatus and a method for generating an omnidirectional signal at the same time as an omnidirectional signal output, And outputting a simultaneous signal having a non-directivity and a non-directivity.

Another object of the present invention is to minimize the distortion of the real-time reproduced signal by using the pre-distortion adaptive filter at the time of outputting the supergain signal and to eliminate the incompleteness of the single sideband filter by using the residual side band modulation, And to provide a composite speaker system which can improve the performance.

It is still another object of the present invention to compare envelope signals of an audio input signal with envelope signals of a distortion compensated signal to which an adaptive filter coefficient of a previous input signal is applied, So as to simplify the design hardware by applying the pre-distortion compensation in real time, and to improve the ultrasonic speaker sound quality.

It is still another object of the present invention to minimize the distortion of a signal that is non-linearly demodulated in air by dynamically modulating the modulation index of the pre-distortion compensated compensation signal during VSB modulation by compensating the distortion according to the level of the input signal And to provide a composite speaker system capable of improving speaker sound quality.

Finally, another object of the present invention is to provide an ultrasonic transducer which generates an inverse filter model of an ultrasonic transducer by applying filtering to an ultrasonic transducer applied to a current system using a specific filter, And it is an object of the present invention to provide a composite speaker system capable of improving sound quality by minimizing distortion when converting a modulated signal into an ultrasonic wave.

The technical problem of the present invention as described above is achieved by the following means.

(1) In a composite speaker system including a supergain speaker system,

A signal input unit for inputting an audio input signal to be amplified; An A / D converter for converting an analog input signal into a digital signal; A signal delay unit delaying the omnidirectional signal by a delay time of a signal output from the supergain speaker system; A pulse width modulator for pulse width modulating the signal output from the signal delay signal unit; And an omnidirectional amplifier for omnidirectionally amplifying and outputting a pulse width modulated signal.

(2) In the above (1), the supergain speaker system comprises:

A first envelope calculation unit for calculating an envelope of the currently input audio input signal;

A square root calculator for calculating a square root of the first envelope signal calculated by the first envelope calculator to generate a square root of a first envelope signal;

A predistortion adaptive filter unit for performing a distortion compensation by applying an adaptive filter coefficient update term according to an adaptive filter coefficient determined at a previous stage to the currently input audio input signal to generate a compensation signal;

A second envelope calculation unit for calculating an envelope of the compensation signal to generate a second envelope signal;

An error arithmetic unit for comparing the second envelope signal and the square root of the first envelope signal to generate an error signal;

An adaptive filter coefficient update unit for calculating an adaptive filter coefficient update term and an adaptive filter coefficient from the error signal;

A dynamic VSB modulator for modulating the compensation signal dynamically into an ultrasonic wave band to generate a modulated signal;

An ultrasound transducer model for modeling an inverse filter corresponding to a frequency characteristic of the ultrasonic transducer and generating a filtering signal by applying the inverse filter to the modulated signal;

An ultrasonic amplifier for amplifying the filtering signal; And

And an ultrasound transducer for converting the amplified signal into an ultrasonic signal.

(3) In the above (2)

Since the signal delay in the signal processing in the first envelope calculation unit, the second envelope calculation unit, the dynamic VSB modulation unit, and the ultrasound transducer model h (t) constituting the supergain speaker system, the supergain signal And outputs a signal delayed by a time equal to a certain delay time.

(4) In the above (2)

If the audio input signal is x (t), the first envelope signal is E (t), and the previous end adaptive filter coefficient update term is a _m (t)

;The compensation signal x (t)

;

The second envelope signal E (t) 'obtained by AM-modulating the compensation signal x (t)' is given by E (t) '= 1 + mx (t)';

The error signal e (t) is: e (t) = E (t) '- E (t) ^0.52 ;

The adaptive filter coefficient update term _{△ a m (t) = -} e (t) / a m (t) = -2E (t) '- E (t) 0.5 x (tm);

Wherein the adaptive filter coefficient is a _m (t + 1) = a _m (t) + 硫 a _m (t)

Wherein m is a modulation index and? Is an adaptation coefficient.

(5) In the above-mentioned (2), the dynamic VSB modulator may comprise:

Wherein the modulation index is dynamically changed according to an input signal level.

(6) In the above (2), the adaptive filter coefficient updating unit may include:

LMS method, and RLS method is applied to a composite speaker system capable of simultaneous signal output of both super-directional and non-directional signals.

(7) In the above (2), the pre-

And a linear FIR filter. ≪ RTI ID = 0.0 > 8. < / RTI >

(8) In the above (2)

Wherein the ultrasonic transducer is preliminarily calculated using the frequency characteristics of the ultrasonic transducer obtained by modeling the ultrasonic transducer with a specific filter.

(9) In the above-mentioned (8)

FIR filter. The composite speaker system is capable of simultaneously outputting a supergain and an omnidirectional signal.

As described above, the composite speaker system capable of simultaneously outputting the supergain signal and the non-directional simultaneous signal according to the embodiment of the present invention allows the non-directional signal output to occur at the same time as the supergain signal output with respect to the input audio signal At the same time, it is possible to output both the omnidirectional signal and the omnidirectional signal.

Further, the composite speaker system according to the present invention minimizes the distortion of the real-time reproduction signal by using the pre-distortion adaptive filter in the process of processing the omnidirectional signal and eliminates the incompleteness of the single sideband filter by using the residual sideband modulation, Can be improved.

The composite speaker system according to an embodiment of the present invention compares an envelope signal of an audio input signal and an envelope signal of a distortion-compensated signal to which an adaptive filter coefficient of a previous input signal is applied, So that the design hardware can be simplified by applying the pre-distortion compensation in real time, and the ultrasonic speaker sound quality can be improved.

The composite speaker system according to another embodiment of the present invention compensates distortion according to the level of the input signal by dynamically modulating the modulation index of the compensation signal with the pre-distortion compensated VSB modulation, So that the speaker sound quality can be improved.

Finally, in the composite speaker system according to another embodiment of the present invention, the ultrasound transducer applied to the current system is filtered with a specific filter, and an inverse filter model of the ultrasonic transducer is generated using the corresponding coefficient values, Signal, it is possible to minimize the distortion in the ultrasonic wave conversion of the modulated signal and improve the sound quality.

FIG. 1 illustrates a method of processing an audio input signal using a square root modulation method in a conventional super-directional speaker system.
2 illustrates a method of processing an audio input signal according to SSB modulation and recursion in a conventional super-directional speaker system.
3 is a diagram of a supergain signal processing system in accordance with an embodiment of the present invention.
4 is a flowchart illustrating a signal processing method of a supergain speaker system according to an embodiment of the present invention.
5 illustrates a non-directional signal output unit according to an embodiment of the present invention.

The above and other objects, features and advantages of the present invention will become more apparent from the following detailed description in conjunction with the accompanying drawings. Hereinafter, embodiments of the present invention will be described in detail with reference to the accompanying drawings.

3 is a diagram illustrating a supergain signal processing system according to an embodiment of the present invention.

Referring to FIG. 3, a supergain speaker system according to an embodiment of the present invention compares an envelope of an audio input signal with an envelope applied with an adaptive filter coefficient obtained from a previous audio input signal, and calculates a current adaptive filter coefficient An adaptive filter calculation unit; A VSB modulator for VSB modulating the audio input signal to which the adaptive filter coefficient value is applied; And an ultrasound converting unit for converting the modulated signal into an ultrasonic wave. The adaptive filter calculating unit includes a first envelope calculating unit 10, a square root calculating unit 20, a second envelope calculating unit 40, an error calculating unit 50, A VSB modulator, a dynamic VSB modulator 70, an ultrasound transducer, an ultrasound transducer model 80, an ultrasound transducer 80, and an ultrasound transducer 80. The pre-distortion adaptive filter 30 includes an adaptive filter coefficient update unit 60, (90), and an ultrasonic transducer (100).

In other words, the supergain speaker system of the present invention includes a first envelope calculation unit 10 for calculating an envelope of a currently input audio input signal x (t) to generate a first envelope signal E (t) for an envelope, A square root calculation unit 20 for calculating an ideal envelope signal E (t) ^0.5 by using the first envelope signal E (t) generated by the first envelope calculation unit 10, a previous-stage audio input signal x (t- 1) by applying an adaptive filter coefficient update term calculated from the envelope of the audio input signal x (t) to generate a distortion compensation signal x (t) ', A second envelope calculation unit 40 for calculating an envelope of the distortion compensated compensation signal x (t) 'output from the pre-distortion adaptive filter unit 30 to generate a second envelope signal E (t)', a E (t) ^0.5 with the second envelope signal E (t) 'of the output signals 20) to the comparison operation An error arithmetic unit 50 for outputting an error signal e (t), an adaptive filter coefficient update unit for calculating a coefficient update term of the pre-distortion adaptive filter corresponding to the error signal e (t) and supplying it to the pre-distortion adaptive filter unit 30 A dynamic VSB modulating unit 70 for modulating the distortion compensated compensation signal x (t) 'output from the predistortion adaptive filter unit 30 into a dynamically ultrasonic band to generate a modulated signal x (t) ) To generate a transformed signal x (t) '''by applying an inverse filter h (t) corresponding to the frequency specific characteristic of the ultrasonic transducer 100 to the modulated signal x (t)'' An ultrasound amplifier 90 for amplifying the converted signal x (t) '''output from the ultrasound transducer model 80 to generate an amplified signal x (t) t) "" into an ultrasonic signal.

Prior to the detailed description, since VSB modulation is a method of symmetrically removing one sideband in AM modulation, a method of approaching by a mathematical expression is similar to AM modulation, and therefore, The VSB modulation is replaced with the AM modulation, and a concrete equation is applied to explain the VSB modulation.

The first envelope calculation unit 10 calculates an envelope for the current audio input signal x (t). The envelope signal E (t) calculated by the first envelope calculation unit 10 can be defined by E (t) which is the same as in Eqs. 1 and 2 described above, and a detailed description thereof will be omitted.

The square root calculator 20 calculates an ideal envelope signal E (t) ^0.5 of the envelope signal E (t) calculated by the first envelope calculator 10. Referring to Equation (1), the mathematically ideal signal of the signal generated by the first envelope calculation unit 10 is a signal corresponding to the square root of the envelope signal E (t). In Equation (1), the second-order partial time derivative can be solved by using a 12 dB / octave equalizer.

The pre-distortion adaptive filter unit 30 applies the adaptive filter coefficient a _m (t) calculated by the previous audio input signal x (t-1) to the currently input audio input signal x (t) And outputs the compensated compensation signal x (t) 'in the same manner.

&Quot; (3) "

 The second envelope calculation unit 40 calculates the envelope signal E (t) 'of the compensation signal x (t)', which is distortion-compensated by the pre-distortion adaptive filter unit 30. The envelope signal E (t) 'calculated by the second envelope calculation unit 40 is obtained after AM modulation of x (t)', and this signal E (t) 'is as shown in Equation (4).

&Quot; (4) "

E (t) '= 1 + mx (t)'

The error calculator 50 subtracts the signal E (t) ^0.5 calculated by the square root calculator 20 from the envelope signal E (t) 'calculated from the second envelope calculator 40 and outputs the error signal e (t ). E (t) calculated by the error arithmetic unit 50 is expressed by Equation (5).

&Quot; (5) "

e (t) = E (t) '- E (t) ^0.52

The adaptive filter coefficient update unit 60 calculates an adaptive filter coefficient update term? A _m (t) by applying an LMS (Least Mean Square) scheme to the error signal e (t) calculated by the error calculation unit 50. The method of calculating the update term? A _m (t) from the error signal e (t) of the present invention can also be applied to the RLS (Recursive Least Square) method. Hereinafter, the LMS method will be mainly described. The update term? A _m (t) calculated by the adaptive filter coefficient update unit 60 can be expressed by Equation (6).

&Quot; (6) "

_{△ a m (t) = -ðe} (t) / ða m (t) = -2E (t) '- E (t) 0.5 x (tm)

Therefore, the adaptive filter coefficient calculated by the adaptive filter coefficient updating unit 60 and supplied to the pre-distortion adaptive filter unit 30 can be expressed by Equation (7).

&Quot; (7) "

_{a m (t + 1) =} a m (t) + β △ a m (t)

However, β is an adaptation coefficient. In the normalized LMS method, β can change steadily with time and converge quickly, and it is possible to design a stable system using β.

Pre-distortion adaptive filter section 30 in real time to update, wherein a _m audio input signal x (t + 1) is input by using the (t + 1) in the next stage obtained by the adaptive filter coefficient update section 60 enemy . The pre-distortion adaptive filter unit 30 may use a linear FIR (Finite Impulse Response) filter to ensure accurate linear phase characteristics.

The dynamic VSB modulation unit 70 dynamically modulates the distortion compensated compensation signal x (t) 'generated by the pre-distortion adaptive filter unit 30 into an ultrasonic band, wherein the signal x (t) VSB modulation is performed to remove most of the lower sideband and to leave the remaining sideband and the remaining sideband complete sideband. In other words, the dynamic VSB modulator 70 dynamically changes the modulation index m according to the signal level of the audio input signal. Since the dynamic VSB modulation removes the signal symmetrically around the carrier frequency, all the information is contained in the remaining spectrum. Therefore, SSB can prevent the deterioration of sound quality caused by demodulation due to incomplete filter characteristics. can do.

The ultrasonic transducer model 80 calculates the inverse filter h (t) according to the ultrasonic transducer 100 and outputs the inverse filter h (t) to the modulated signal x (t) "generated by the dynamic VSB modulator 70 (T) '' 'from the frequency characteristics of the ultrasound transducer 100 when the ultrasound transducer 100 is modeled by a specific filter, for example, an FIR filter. The coefficients of the filter can be obtained and the inverse filter coefficients can be obtained in advance using the coefficients of the obtained filters.

The ultrasonic wave amplifier 90 converts the ultrasonic wave generated by the ultrasonic vibration element into the signal x (t) '' 'obtained by filtering the signal x (t) modulated by the dynamic VSB modulation unit 70 with the inverse filter h (T) '' 'by oscillating the signal with a physical force by radiating a signal x (t)' '' to produce an amplitude amplified signal x (t) '' '.

The ultrasonic transducer 100 converts the amplitude amplified signal x (t) "" by the ultrasonic amplifier 90 into an ultrasonic signal. The ultrasonic transducer 100 may have a piezoelectricity, a self-distortion, or a semiconductor.

A piezoelectric electroacoustic transducer is a transducer that uses ultrasonic waves generated from crystals when a high frequency voltage of a suitable frequency is applied to a plate or rod cut in a certain direction in a crystal such as crystal. This piezoelectric electroacoustic transducer utilizes an interference phenomenon which occurs when the frequency of the applied voltage becomes an odd multiple of the fundamental frequency of crystal determination. That is, in order to obtain a specific frequency, the piezoelectric electroacoustic transducer applies an appropriate vibrator, for example, a vibration suitable for quartz, and is called a piezoelectric resonator because it causes vibration by applying electricity.

The principle of generating the ultrasonic wave in the self-distorting type or the semiconductor type is the same as that of the piezoelectric type, and it is classified into the difference in the characteristics of the material to be used.

The ultrasonic signal converted by the ultrasonic transducer 100 is radiated into the air to be nonlinearly demodulated and output as acoustic audio.

A signal processing method according to a distortion compensation method of a supergain speaker system according to an embodiment of the present invention will now be described with reference to FIG.

Before describing, x (t) is the currently input audio input signal, and h (t) is an inverse filter of the coefficient values calculated by modeling various ultrasound transducers 100 with a specific filter.

The signal processing method of the supergain speaker system of the present invention first calculates the envelope of the currently input audio input signal x (t) (S1), performs a square root operation of the calculated envelope signal E (t) t) ^0.5 (S2).

Meanwhile, during the steps S1 and S2, the current audio input signal x (t) is obtained by applying the adaptive filter coefficient calculated from the audio input signal x (t-1) of the previous stage to the distortion compensated signal x (S3), and calculates an envelope signal E (t) 'of the generated x (t)' signal (S4).

Next, the signals E (t) ^0.5 and E (t) 'calculated in steps S2 and S4 are calculated (S5).

E (t) ^0.5 , which is an ideal envelope signal, is subtracted from the envelope signal E (t) 'to generate the error signal e (t).

Next, the adaptive filter coefficient update unit 60 calculates an update term according to the signal e (t) (S6).

In order to calculate the update term, the adaptive filter coefficient update unit 30 uses at least one adaptive scheme of LMS (Least Mean Square) and RLS scheme.

Next, using the update term of the e (t) signal, the audio input signal x (t + 1) input at the next stage is pre-distorted compensated (S3).

In step S3, a distortion compensated signal x (t) 'to which the adaptive filter coefficient calculated by the previous stage audio input signal x (t-1) is applied is dynamically VSB-modulated to generate a signal x (t) .

Next, the inverse filter h (t) of the ultrasonic transducer model is applied to the VSB modulated signal x (t) "(S8).

In this case, the inverse filter h (t) to be applied may be calculated by modeling the ultrasonic transducer 100 used in the system with a specific filter.

Next, the ultrasonic amplifier 90 ultrasonically amplifies the signal x (t) '' 'filtered by the inverse filter h (t) (S9).

Next, the ultrasonic transducer 100 converts the amplified signal into ultrasonic waves (S10).

Finally, the ultrasound signal is nonlinearly demodulated in air and output as an audio audio signal out (t) (S11).

The supergain speaker system according to an embodiment of the present invention provides a modulation unit with a pre-distortion-compensated signal using an adaptive filter, so that distortion compensation can be applied in real time without repetition. Therefore, the supergain speaker system of the present invention can reduce the delay due to the distortion compensation, simplify the hardware design, and facilitate the construction of a system capable of efficient modulation.

That is, the supergain speaker system according to an embodiment of the present invention compensates for distortion of the audio input signal in real time by using the pre-distortion adaptive filtering, and the audible signal, which is reproduced secondarily from the ultrasonic transducer, It is possible to pre-distort before modulation to be close to the signal, and using a linear FIR filter allows the pre-distorted signal to be distorted within its original bandwidth, and even in hardware design, the complexity is low. In addition, since the supergain speaker system of the present invention uses the VSB modulation scheme, the information about the low frequency band of the original signal is preserved without being overlapped by the symmetrical filter, so that the SSB modulation The modulation quality can be improved by changing the modulation index dynamically according to the level of the input signal.

The speaker system according to the present invention includes an omnidirectional signal output as shown in Fig. 5 to output an omnidirectional signal. The non-directional signal output unit according to the present invention may be physically separated or integrated with the supergain signal processing system shown in FIG.

The omni-directional signal output unit includes a signal input unit 200, an A / D converter 210, a signal delay unit 220, a pulse width modulator 230, and an omni-directional amplifier 240.

When the audio input signal x (t) is input to the signal input unit 200, the A / D converter 210 converts the analog input signal x (t) into a digital signal. The output of the signal converted into the digital signal is delayed in the signal delay unit 220. The delay time of the output of the signal is delayed by the first envelope calculation unit 10 in the supergain signal processing system, Directional signal is delayed by the same amount of time as that of the directional signal due to the signal processing in the unit 40, the dynamic VSB modulation 70, and the ultrasonic transducer model h (t) 80.

Therefore, the signal delay unit 220 can perform the signal processing in the first envelope calculation unit 10, the second envelope calculation unit 40, the dynamic VSB modulation 70, and the ultrasonic transducer model h (t) It is preferable to perform adaptive signal processing so as to output a delay time corresponding to the same delay time with reference to the delay time caused thereby.

5, the signal delayed by the time delay is output to the non-directional signal output of the audio signal input through the pulse width modulator 230 and the omnidirectional amplifier 240, So that it can occur at the same time as the signal output so that both the supergain signal and the non-directional signal can be output at the same time.

As described above, each block of the block diagrams attached hereto and combinations of steps of the flowchart diagrams may be performed by the program instructions. These program instructions may be embedded in a processor of a general purpose digital terminal, or other programmable data processing apparatus, so that the instructions performed through these processors may be used to implement the functions described in each block or flowchart of the block diagram Respectively. These program instructions may also be stored in a digital terminal available or digital terminal readable memory capable of directing a computer or other programmable data processing equipment to implement a function in a particular manner, It is also possible for the instructions stored in the readable memory to produce an article of manufacture containing instruction means for performing the function described in each block or flowchart of each block diagram. Program instructions may also be loaded on a digital terminal or other programmable data processing equipment so that a series of operating steps may be performed on a digital terminal or other programmable data processing equipment to create a process running on the digital terminal to generate a digital terminal or other It is also possible that the instructions that perform the programmable data processing equipment provide the steps for executing the functions described in each block of the block diagram and at each step of the flowchart.

Also, each block or each step may represent a module, segment, or portion of code that includes one or more executable instructions for executing the specified logical function (s). It should also be noted that in some alternative embodiments, the functions mentioned in the blocks or steps may occur out of order. For example, two blocks or steps shown in succession may in fact be performed substantially concurrently, or the blocks or steps may sometimes be performed in reverse order according to the corresponding function.

The foregoing description is merely illustrative of the technical idea of the present invention, and various changes and modifications may be made by those skilled in the art without departing from the essential characteristics of the present invention. Therefore, the embodiments disclosed in the present invention are intended to illustrate rather than limit the scope of the present invention, and the scope of the technical idea of the present invention is not limited by these embodiments. The scope of protection of the present invention should be construed according to the following claims, and all technical ideas falling within the scope of the same shall be construed as falling within the scope of the present invention.

10, 40: envelope calculation unit
20:
30: a predistortion adaptive filter unit
50:
60: adaptive filter coefficient update unit
70: Dynamic VSB modulation section
80: Ultrasonic transducer model
90: Ultrasonic amplifier
100: Ultrasonic transducer
200: Signal input section
210: A / D converter
220: Signal delay unit
230: Pulse width modulator
240: Omnidirectional amplifier

Claims (9)

A first envelope calculation unit for calculating an envelope of the currently input audio input signal;
A square root calculator for calculating a square root of the first envelope signal calculated by the first envelope calculator to generate a square root of a first envelope signal;
A predistortion adaptive filter unit for performing a distortion compensation by applying an adaptive filter coefficient update term according to an adaptive filter coefficient determined at a previous stage to the currently input audio input signal to generate a compensation signal;
A second envelope calculation unit for calculating an envelope of the compensation signal to generate a second envelope signal;
An error arithmetic unit for comparing the second envelope signal and the square root of the first envelope signal to generate an error signal;
An adaptive filter coefficient update unit for calculating an adaptive filter coefficient update term and an adaptive filter coefficient from the error signal;
A dynamic VSB modulator for modulating the compensation signal dynamically into an ultrasonic wave band to generate a modulated signal;
An ultrasound transducer model for modeling an inverse filter corresponding to a frequency characteristic of the ultrasonic transducer and generating a filtering signal by applying the inverse filter to the modulated signal;
An ultrasonic amplifier for amplifying the filtering signal; And
And a supergain speaker system including the ultrasonic transducer for converting the amplified signal into an ultrasonic signal,
A signal input unit for inputting an audio input signal to be amplified;
An A / D converter for converting an analog input signal into a digital signal;
A signal delay for delaying and outputting the omnidirectional signal by the delay time of the signal caused by the signal processing in the first envelope calculation unit, the second envelope calculation unit, the dynamic VSB modulation unit, and the ultrasonic transducer model of the supergain speaker system; part;
A pulse width modulator for pulse width modulating the signal output from the signal delay unit; And
Composite speaker system capable of omnidirectional and omnidirectional simultaneous signal output, including omnidirectional amplifier for omnidirectionally amplifying and outputting pulse width modulated signals. delete delete The method according to claim 1,
If the audio input signal is x (t), the first envelope signal is E (t), and the adaptive filter coefficient update term is a _m (t)
The compensation signal x (t)

;
The second envelope signal E (t) 'obtained by AM-modulating the compensation signal x (t)' is given by E (t) '= 1 + mx (t)';
The error signal e (t) is: e (t) = E (t) '- E (t) ^0.52 ;
The adaptive filter coefficient update term _{△ a m (t) = -} e (t) / a m (t) = -2E (t) '- E (t) 0.5 x (tm);
Wherein the adaptive filter coefficient is a _m (t + 1) = a _m (t) + 硫 a _m (t)
Wherein m is a modulation index and? Is an adaptation coefficient. The apparatus of claim 4, wherein the dynamic VSB modulator comprises:
Wherein the modulation index is dynamically changed according to an input signal level. 2. The apparatus of claim 1, wherein the adaptive filter coefficient updating unit comprises:
LMS method, and RLS method is applied to a composite speaker system capable of simultaneous signal output of both super-directional and non-directional signals. 2. The apparatus of claim 1, wherein the predistortion adaptive filter unit comprises:
And a linear FIR filter. ≪ RTI ID = 0.0 > 8. < / RTI > The apparatus of claim 1,
Wherein the ultrasonic transducer is preliminarily calculated using the frequency characteristics of the ultrasonic transducer obtained by modeling the ultrasonic transducer with a specific filter. 9. The method of claim 8,
FIR filter. The composite speaker system is capable of simultaneously outputting a supergain and an omnidirectional signal.

Images