Ultra-directional speaker system and signal processing method

Abstract

The present invention relates to an ultra-directional speaker system comprising: a first envelope calculation unit for calculating the envelope of the currently input audio input signal; a square root calculation unit for generating a square root signal of a first envelope signal by calculating a square root of the first envelope signal calculated by the first envelope calculation unit; a pre-distortion adaptive filter unit for generating a compensation signal by performing distortion compensation in which an adaptive filter coefficient update term according to an adaptive filter coefficient determined in a previous stage is applied to the currently input audio input signal; a second envelope calculation unit for generating a second envelope signal by calculating the envelope of the compensation signal; an error calculation unit for generating an error signal by comparing and calculating the second envelope signal with the square root signal of the first envelope 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 signal modulation unit for generating a modulation signal by dynamically modulating the compensation signal into an ultrasonic band; an ultrasonic transducer model for modeling an inverse filter corresponding to frequency characteristics of an ultrasonic transducer, and generating a filtering signal by applying the inverse filter to the modulation signal; an ultrasonic amplifier for amplifying the filtering signal; and the ultrasonic transducer for converting the amplified signal into an ultrasonic signal, wherein the dynamic signal modulation unit includes: an AM signal modulation unit for generating an AM modulation signal of transmission data; a first sideband signal modulation unit, which receives the AM modulation signal and performs a Hilbert operation thereon so as to obtain a first sideband modulation signal from which a carrier wave has been removed; and a second sideband signal unit, which receives the first sideband modulation signal and adds the carrier wave thereto so as to obtain a second sideband modulation signal.

Description

Super-directional speaker system and signal processing method

[0001] The present invention relates to an ultrasonic speaker that reproduces ultra-directional audio, and more particularly to an ultra-directional speaker system and a signal processing method to which a new signal processing method capable of improving the sound quality of the ultrasonic speaker system is applied.

[0002] Generally, speakers generate sound by converting electrical signals into vibrations and transmitting them through the air. These speakers transmit vibrations in an isotropic manner. Consequently, listeners can hear the sound emanating from the speaker from all directions relative to it. However, this isotropy of speakers can sometimes cause unnecessary problems. For instance, when an installation such as an art gallery or museum houses various artworks or exhibits and is designed to provide explanations via speakers, interference occurs between the sounds generated by the speakers due to the confined space of the venue. Furthermore, if a large number of people simultaneously listen to explanations for different artworks and exhibits, the volume of audio guidance interferes with and is distorted, turning into significant noise. To address these issues, super-directional speakers have emerged, capable of reproducing sound so that it can be heard only from a specific direction.

[0003] One of the conventional super-directional speaker methods is the use of a parabolic dish. A parabolic super-directional speaker is designed so that a standard speaker is installed at the focal point of a parabolic dish, causing the speaker's acoustic output to reflect off the dish and travel in a straight line. Because this type of speaker is frequently used in places like museums, it is also widely known as a museum speaker. However, this conventional super-directional speaker method using a parabolic dish has several drawbacks: the diameter of the dish is considerably large, the distance over which sound is transmitted with directionality is short (less than 10m), and the sound quality is significantly poor, making it very difficult to achieve satisfactory characteristics.

[0004] Accordingly, ultrasonic speaker technology utilizing the nonlinear interference phenomenon that ultrasound causes with air is being widely applied in the implementation of super-directional speakers. Although ultrasonic speaker technology has been under development since the 1960s, commercialization was delayed due to the sluggish development of various peripheral devices and issues regarding industrial profitability, but it is now being developed in earnest.

[0005] A supercardioid speaker system consists of a signal processing unit for obtaining appropriate sound quality, a modulation unit for efficiently modulating the processed signal into the ultrasonic band, an ultrasonic amplifier for driving the ultrasonic transducer, and an ultrasonic transducer that actually generates ultrasound 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 derivative can be solved using an equalizer of 12 dB / octave, and the resulting envelope E(t) can be expressed as shown in Equation 2.

[0006]

[0007]

[0008] where m is the modulation index and x(t) is the original audible audio signal.

[0009] In the above mathematical formula, if the audible signal p(t) heard through the ultrasonic speaker is proportional to the original audible audio signal x(t), it is possible to reproduce audible sound without distortion. However, in reality, as shown in Equation 1, severe distortion corresponding to the square of the original signal x(t) occurs. As a method to reduce this distortion, conventional ultrasonic speakers can reduce distortion by decreasing the modulation index m, but this lowers the reproduction efficiency, making it difficult to obtain high acoustic output.

[0010] A distortion compensation method to compensate for other distortions is to modulate the square root of the original signal, as shown in Fig. 1. Theoretically, according to this method, the original signal can be faithfully reproduced, but due to the non-linear operation of the square root, the spectrum of the original signal x(t), whose bandwidth is limited, appears on a bandwidth of almost infinite. Therefore, without an ultrasonic transducer that reproduces an infinite bandwidth, the ultrasonic speaker of the method shown in Fig. 1 is bound to have an absolute limit in reducing distortion.

[0011] To solve the problem illustrated in Fig. 1, ATC of the United States presented an iterative error compensation method without increasing bandwidth in its patent application titled "Modulator Processing for a Parametric Speaker System (US 6,584,205)" illustrated in Fig. 2. Briefly, the patent invented by ATC of the United States is a method that compensates for sound quality distortion by calculating an ideal modulation signal waveform through a Single Side Band (SSB) channel model without a converter, comparing it with the actual modulated signal to calculate the error, and repeating the process of compensating this error for the signal before modulation several times. However, since the invention patent by ATC of the United States compensates for errors iteratively, such iterative error compensation inevitably generates a large amount of computation. Consequently, not only does this make the hardware design very complex, but it also has the serious disadvantage of increasing delays due to signal processing. Furthermore, the invention patent by ATC of the United States applies SSB modulation, but to avoid distortion caused by the imperfections of the SSB filter, it has the problem of requiring a significantly sharp SSB filter design by increasing the order of the SSB filter.

[0012] The present invention aims to solve the above problems, and the objective of the present invention is to provide a super-directional speaker system and a signal processing method capable of improving sound quality by minimizing distortion of a real-time playback signal using a pre-distortion adaptive filter and eliminating the imperfections of a single-sideband filter using improved dynamic signal modulation.

[0013] Another objective of the present invention is to provide a super-directional speaker system and a signal processing method that can simplify design hardware and improve ultrasonic speaker sound quality by applying prior distortion compensation in real time, by comparing the envelope signal of an audio input signal with the envelope signal of a distortion-compensated signal to which adaptive filter coefficients of a previous input signal are applied, and calculating and applying the adaptive filter coefficient values ​​of the current input signal accordingly.

[0014] Another objective of the present invention is to provide a super-directional speaker system and a signal processing method capable of improving speaker sound quality by minimizing distortion of a signal non-linearly demodulated in air by dynamically modulating the modulation index of a pre-distortion-compensated compensation signal during the signal modulation process and compensating for distortion according to the level of the input signal.

[0015] Finally, another objective of the present invention is to provide a super-directional speaker system and a signal processing method capable of improving sound quality by minimizing distortion during ultrasonic conversion of a modulated signal, by filtering an ultrasonic transducer currently applied to a system with a specific filter, generating an inverse filter model of the ultrasonic transducer using the corresponding coefficient values, and applying it to a dynamically modulated signal.

[0016] To achieve the above objective, a super-directional speaker system according to an embodiment of the present invention comprises: a first envelope calculation unit that calculates the envelope of a currently input audio input signal; a square root calculation unit that generates a square root signal of the first envelope signal by calculating the square root of the first envelope signal calculated by the first envelope calculation unit; a pre-distortion adaptive filter unit that generates a compensation signal by performing distortion compensation by applying an adaptive filter coefficient update term according to the adaptive filter coefficients determined in the previous stage to the currently input audio input signal; a second envelope calculation unit that generates a second envelope signal by calculating the envelope of the compensation signal; an error calculation unit that generates an error signal by comparing the second envelope signal with the square root signal of the first envelope signal; an adaptive filter coefficient update unit that calculates an adaptive filter coefficient update term and adaptive filter coefficients from the error signal; and a dynamic signal modulation unit that generates a modulated signal by dynamically modulating the compensation signal into an ultrasonic band. The ultrasonic transducer model that models an inverse filter corresponding to the frequency characteristics of the ultrasonic transducer and applies it to the modulation signal to generate a filtered signal; an ultrasonic amplifier that amplifies the filtered signal; and the ultrasonic transducer that converts the amplified signal into an ultrasonic signal are provided.

[0017] A super-directional speaker system according to an embodiment of the present invention is characterized by comprising: an adaptive filter calculation unit that calculates current adaptive filter coefficients by comparing the envelope of a current audio input signal with an envelope to which adaptive filter coefficients obtained from a previous stage audio input signal are applied; a dynamic signal modulation unit that dynamically modulates the audio input signal to which the adaptive filter coefficient values ​​are applied; and an ultrasonic converter that converts the modulated signal into ultrasonic waves.

[0018] A super-directional speaker signal processing method according to an embodiment of the present invention comprises: a first step of generating a first envelope signal by calculating the envelope of a current audio input signal; a second step of generating an ideal envelope signal of the first envelope signal; a third step of generating a pre-distortion compensated signal by applying adaptive filter coefficients determined by the previous audio input signal; a fourth step of generating an envelope signal of the compensated signal; a fifth step of generating an error signal by performing a comparison operation between the ideal envelope signal and the envelope signal of the compensated signal; a sixth step of calculating an adaptive filter coefficient update term and an adaptive filter coefficient from the error signal; a seventh step of generating a modulated signal by dynamically modulating the compensated signal; an eighth step of filtering the modulated signal with an inverse filter corresponding to an ultrasonic transducer; a ninth step of ultrasonically amplifying the filtered signal; and a tenth step of ultrasonically converting the ultrasonically amplified signal.

[0019] The super-directional speaker system and signal processing method according to an embodiment of the present invention can improve sound quality by minimizing distortion of the real-time playback signal using a pre-distortion adaptive filter and eliminating the imperfections of the single-sideband filter using an improved dynamic signal modulation method.

[0020] A super-directional speaker system and signal processing method according to an embodiment of the present invention can simplify design hardware and improve ultrasonic speaker sound quality by applying prior distortion compensation in real time, by comparing the envelope signal of an audio input signal with the envelope signal of a distortion-compensated signal to which adaptive filter coefficients of a previous input signal are applied, and by calculating and applying the adaptive filter coefficient values ​​of the current input signal accordingly.

[0021] A super-directional speaker system and signal processing method according to another embodiment of the present invention can improve speaker sound quality by minimizing distortion of a signal that is non-linearly demodulated in air, by dynamically modulating the modulation index of a pre-distortion-compensated compensation signal during dynamic signal modulation, thereby compensating for distortion according to the level of the input signal.

[0022] Finally, the super-directional speaker system and signal processing method according to another embodiment of the present invention can improve sound quality by minimizing distortion during ultrasonic conversion of the modulated signal by filtering the ultrasonic transducer currently applied to the system with a specific filter and generating an inverse filter model of the ultrasonic transducer using the coefficient values ​​and applying it to the VSB modulated signal.

[0023] In addition, preferred embodiments of the present invention are disclosed for illustrative purposes only, and those skilled in the art may make various modifications, changes, and additions within the spirit and scope of the present invention, and such modifications and changes should be considered to fall within the scope of the following claims.

[0024] FIG. 1 is a diagram illustrating an audio input signal processing method using a square root modulation scheme in a conventional super-directional speaker system.

[0025] FIG. 2 is a diagram illustrating an audio input signal processing method based on SSB modulation and recursion in a conventional super-directional speaker system.

[0026] FIG. 3 is a drawing showing a super-directional speaker system according to an embodiment of the present invention.

[0027] FIG. 4 is a flowchart illustrating a signal processing method of a super-directional speaker system according to an embodiment of the present invention.

[0028] FIG. 5 is a configuration diagram showing a dynamic signal modulation unit according to an embodiment of the present invention.

[0029] FIG. 6 is an explanatory diagram showing the process of signal modulation in a dynamic signal modulation unit according to an embodiment of the present invention.

[0030] FIG. 7 is an example configuration and explanatory diagram of a speaker unit according to an embodiment of the present invention.

[0031] The advantages and features of the present invention and the methods for achieving them will become clear by referring to the embodiments described below in detail together with the accompanying drawings. However, the present invention is not limited to the embodiments disclosed below but can be implemented in various different forms. These embodiments are provided merely to ensure that the disclosure of the present invention is complete and to fully inform those skilled in the art of the scope of the invention, and the present invention is defined only by the scope of the claims.

[0032] Throughout the specification, when a part is described as "comprising" or "including" a component, this means that, unless specifically stated otherwise, it does not exclude other components but may include additional components. Furthermore, the term "...part" as used in the specification refers to a unit that performs at least one function or operation. Additionally, "one (a or an)," "one," "the," and similar related terms may be used in the context describing the invention (particularly in the context of the following claims) to include both singular and plural forms, unless otherwise indicated in the specification or clearly contradicted by the context.

[0033] In describing the embodiments of the present invention, specific descriptions of known functions or configurations will be omitted if it is determined that such detailed descriptions could unnecessarily obscure the essence of the invention. Furthermore, the terms described below are defined in consideration of their functions in the embodiments of the present invention, and these definitions may vary depending on the intentions or practices of the user or operator. Therefore, such definitions should be based on the content throughout this specification.

[0034] Hereinafter, embodiments of the present invention will be examined with reference to the attached drawings.

[0035] FIG. 3 is a drawing illustrating a super-directional speaker system according to an embodiment of the present invention.

[0036] Referring to FIG. 3, a super-directional speaker system according to an embodiment of the present invention comprises: an adaptive filter calculation unit that calculates current adaptive filter coefficients by comparing the envelope of a current audio input signal with an envelope to which adaptive filter coefficients obtained from a previous audio input signal are applied; a dynamic signal modulation unit that dynamically modulates an audio input signal to which the adaptive filter coefficient values ​​are applied; and an ultrasonic converter that converts the modulated signal into ultrasonic waves. The adaptive filter calculation unit includes a first envelope calculation unit (10), a square root calculation unit (20), a second envelope calculation unit (40), an error calculation unit (50), and an adaptive filter coefficient update unit (60), and includes a pre-distortion adaptive filter unit (30) for applying adaptive filter coefficients.

[0037] In other words, the super-directional speaker system of the present invention calculates the envelope of a currently input audio input signal x(t) and generates a first envelope signal E(t) for the envelope, and uses the first envelope signal E(t) generated by the first envelope calculation unit (10) to obtain an ideal envelope signal E(t). 0.5 A square root operation unit (20) that calculates, a pre-distortion adaptive filter unit (30) that generates a distortion-compensated signal x(t)' by performing pre-distortion compensation of the current audio input signal x(t) by applying an adaptive filter coefficient update term calculated from the envelope of the previous audio input signal x(t-1), a second envelope calculation unit (40) that generates a second envelope signal E(t)' by calculating the envelope of the distortion-compensated compensation signal x(t)' output from the pre-distortion adaptive filter unit (30), and E(t), which is the output signal of the square root operation unit (20). 0.5The system comprises an error calculation unit (50) that compares and calculates the second envelope signal E(t)' and outputs the error signal e(t); an adaptive filter coefficient update unit (60) that calculates the coefficient update term of a pre-distortion adaptive filter corresponding to the error signal e(t) and supplies it to a pre-distortion adaptive filter unit (30); a dynamic signal modulation unit (70) that dynamically modulates the distortion-compensated compensation signal x(t)' output from the pre-distortion adaptive filter unit (30) into an ultrasonic band to generate a modulated signal x(t)"; an ultrasonic transducer model (80) that generates a converted signal x(t)''' by modeling an inverse filter h(t) corresponding to the frequency-specific characteristics of the ultrasonic transducer (100) and applying it to the modulated signal x(t)"; an ultrasonic amplifier (90) that amplifies the converted signal x(t)''' output from the ultrasonic transducer model (80) to generate an amplified signal x(t)""; and an ultrasonic transducer (100) that converts the output of the ultrasonic-amplified amplified signal x(t)"" into an ultrasonic signal.

[0038] Before providing a detailed description, the dynamic signal modulation unit (70) according to the present invention includes an AM signal modulation unit (71) and an improved SSB signal modulation unit (73) as shown in FIG. 5, wherein the improved SSB signal modulation unit (73) includes a first sideband signal modulation unit (73a) and a second sideband signal modulation unit (73b).

[0039] The above AM signal modulation unit (71) represents typical amplitude modulation in which both wave bands and a carrier wave exist, and modulates digital voice data into an analog signal as shown in Equation (1) below.

[0040] (1)

[0041] At this time, is a sound wave signal, and m is the 'modulation level' of AM modulation.

[0042] The first sideband signal modulation unit (73a) performs a sideband signal generation operation to obtain only one sideband signal (upper sideband or lower sideband) from which the carrier wave has been removed. At this time, the sideband signal generation operation is preferably a Hilbert operation, and signal modulation is performed as shown in Equation (2) below. As shown in FIG. 6, the AM modulated signal modulated by the first sideband signal modulation unit (73a) is generated as a signal in which the carrier wave has been removed and the envelope has been destroyed.

[0043] (2)

[0044] In the above equation (2), is a Hilbert transformed sound wave signal, , It is displayed as, is a carrier wave. silver With the Hilbert transformed sound wave signal, When am.

[0045] The second sideband signal modulation unit (73b) of the present invention adds a carrier wave to the carrier-removed SSB modulation signal obtained from the first sideband signal modulation unit (73a) to generate a complete SSB modulation signal as shown in Equation (3) below.

[0046] (3)

[0047] The modulation signal obtained from the second sideband signal modulation unit (73b) above is converted into a complete SSB modulation signal, and the carrier wave As a result of adding [this], as shown in Fig. 6, the envelope is reconstructed, and unlike conventional VSB modulation, an SSB modulated signal is obtained in which one sideband is completely removed. In addition, the problem of high computational requirements in conventional SSB modulation can be easily solved with hardware acceleration alone, and hardware acceleration can be easily implemented as modern DSP processors have a built-in FFT accelerator.

[0048] Since the dynamic signal modulation according to the present invention is similar to AM modulation in its approach using mathematical formulas, in order to effectively explain the signal conversion process of the super-directional speaker system according to the embodiment of the present invention, the dynamic signal modulation according to the present invention will be substituted with the case of AM modulation and explained by applying specific mathematical formulas.

[0049] The first envelope calculation unit (10) calculates the 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 as the same E(t) as in the previously described mathematical formulas 1 and 2, so a detailed explanation is omitted.

[0050] The square root operation unit (20) is the ideal envelope signal E(t) of the envelope signal E(t) calculated by the first envelope calculation unit (10) 0.5 Calculate. Referring to Equation 1, the mathematically most ideal signal of the signal generated in the first envelope calculation unit (10) is the signal corresponding to the square root of the envelope signal E(t). In Equation 1, the second-order time partial derivative can be solved using a 12 dB / octave equalizer.

[0051] The pre-distortion adaptive filter section (30) has adaptive filter coefficients a calculated by the previous stage audio input signal x(t-1) on the currently input audio input signal x(t). m By applying (t), a distortion-compensated signal x(t)' is output in the same manner as Equation 3.

[0052]

[0053] The second envelope calculation unit (40) calculates the envelope signal E(t)' of the compensation signal x(t)', which is distorted and compensated by the prior distortion adaptation 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.

[0054]

[0055] The error calculation unit (50) calculates the signal E(t) calculated by the square root calculation unit (20) from the envelope signal E(t)' calculated by the second envelope calculation unit (40). 0.5 The error signal e(t) is generated by subtracting it. The e(t) calculated by the error calculation unit (50) is equal to Equation 5.

[0056]

[0057] The adaptive filter coefficient update unit (60) applies the Least Mean Square (LMS) method to the error signal e(t) calculated by the error calculation unit (50) to the adaptive filter coefficient update term △a m Calculate (t). From the error signal e(t) of the present invention, the update term △a m The Recursive Least Square (RLS) method can also be applied to calculate (t). Below, the explanation will focus on the LMS method. Update term △a calculated by the adaptive filter coefficient update unit (60). m (t) can be expressed as in mathematical equation 6.

[0058]

[0059] Accordingly, the adaptive filter coefficients calculated by the adaptive filter coefficient update unit (60) and supplied to the pre-distortion adaptive filter unit (30) can be expressed as Equation 7.

[0060]

[0061] However, β is an adaptation factor. In the normalized LMS method, β varies over time and can converge quickly and stably, and stable system design is possible using β.

[0062] The pre-distortion adaptive filter unit (30) is an update term a obtained by the adaptive filter coefficient update unit (60). mUsing (t+1), the audio input signal x(t+1) input to the next stage is applied in real time. The pre-distortion adaptive filter section (30) may use a linear FIR (Finite Impulse Response) filter to secure accurate linear phase characteristics.

[0063] The dynamic signal modulation unit (70) dynamically modulates the distortion-compensated signal x(t)' generated by the pre-distortion adaptive filter unit (30) into the ultrasonic band, at which time the signal x(t)' completely removes a portion of the upper or lower sideband. In other words, the dynamic signal modulation unit (70) dynamically changes the modulation index m according to the signal level of the audio input signal. Since dynamic signal modulation removes the signal symmetrically around the carrier frequency, all information is contained in the remaining spectrum, so it can prevent the sound quality degradation that occurs during demodulation due to some information being lost or superimposed because of the incomplete filter characteristics of the existing SSB.

[0064] The ultrasonic transducer model (80) calculates an inverse filter h(t) according to the ultrasonic transducer (100) and applies this inverse filter h(t) to the modulated signal x(t) generated by the dynamic signal modulation unit (70) to generate the signal x(t)'''. When the ultrasonic transducer (100) is modeled as a specific filter, for example, an FIR filter, the coefficients of the filter can be obtained from the frequency characteristics of the ultrasonic transducer (100), and the coefficients of the inverse filter can be obtained in advance using the obtained coefficients of the filters.

[0065] The ultrasonic amplifier (90) generates an amplitude-enlarged signal x(t)" by radiating ultrasonic waves generated by an ultrasonic vibration element onto a signal x(t)''' that has been filtered by an inverse filter h(t) of a signal x(t)" modulated by a dynamic signal modulation unit (70), thereby vibrating the signal with physical force to amplify the amplitude of the signal x(t)'''.

[0066] The ultrasonic transducer (100) converts the amplitude-amplified signal x(t)" by the ultrasonic amplifier (90) into an ultrasonic signal. Such ultrasonic transducers (100) can be of the type that utilizes piezoelectricity, magnetic distortion, and semiconductors.

[0067] A piezoelectric electroacoustic transducer is a transducer that utilizes the principle in which ultrasound is generated from a crystal when a high-frequency voltage of an appropriate frequency is applied to a plate or rod cut in a specific direction from a crystal, such as quartz. This piezoelectric electroacoustic transducer utilizes the interference phenomenon that occurs when the frequency of the applied voltage is an odd multiple of the fundamental frequency of the quartz crystal. In other words, to obtain a specific frequency, a suitable vibrator—such as a quartz crystal—is applied; since it induces vibration by applying electricity, it is called a piezoelectric device.

[0068] The principle by which magnetic distortion and semiconductor types generate ultrasound is the same as that of the piezoelectric type, and they are distinguished by differences in the characteristics of the materials used.

[0069] The ultrasonic signal converted by the ultrasonic transducer (100) is radiated into the air and non-linearly demodulated to be output as acoustic audio.

[0070] A signal processing method according to the distortion compensation method of a super-directional speaker system according to an embodiment of the present invention having such a structure is described as follows with reference to FIG. 4.

[0071] Before explaining, x(t) is the currently input audio input signal, and h(t) is the inverse filter of coefficient values ​​calculated by modeling various ultrasonic transducers (100) as specific filters.

[0072] The signal processing method of the super-directional speaker system of the present invention first calculates the envelope of the currently input audio input signal x(t) (S1), and performs a square root operation on the calculated envelope signal E(t) to obtain the signal E(t).0.5 Creates (S2).

[0073] Meanwhile, while steps S1 and S2 are in progress, the current audio input signal x(t) is given a distortion-compensated signal x(t)' by applying adaptive filter coefficients calculated from the previous audio input signal x(t-1) (S3), and the envelope signal E(t)' of the generated x(t)' signal is calculated (S4).

[0074] Next, the signal E(t) calculated in steps S2 and S4, respectively 0.5 , E(t)' is computed (S5).

[0075] From the envelope signal E(t)', the ideal envelope signal E(t) 0.5 Subtract to generate the error signal e(t).

[0076] Next, the adaptive filter coefficient update unit (60) calculates an update term according to the e(t) signal (S6).

[0077] To calculate the update term, the adaptive filter coefficient update unit (30) uses at least one adaptive method among LMS (Least Mean Square) and RLS methods.

[0078] Next, the audio input signal x(t+1) input to the next stage is pre-distorted using the update term of the e(t) signal (S3).

[0079] In step S3, a distortion compensation signal x(t)' with adaptive filter coefficients calculated by the previous audio input signal x(t-1) is dynamically modulated to generate a signal x(t)” (S7).

[0080] Next, the inverse filter h(t) of the ultrasonic transducer model is applied to the dynamically modulated signal x(t) (S8).

[0081] At this time, the applied inverse filter h(t) can be calculated by modeling the ultrasonic transducer (100) used in the system as a specific filter.

[0082] Next, the ultrasonic amplifier (90) ultrasonically amplifies the signal x(t)''' filtered by the inverse filter h(t) (S9).

[0083] Next, the ultrasonic transducer (100) converts the amplified signal into an ultrasonic signal (S10).

[0084] Finally, the ultrasonic signal is non-linearly demodulated in air and output as an acoustic audio signal out(t) (S11).

[0085] The super-directional speaker system according to this embodiment of the present invention provides a pre-distortion-compensated signal to the modulator using an adaptive filter, thereby enabling the application of distortion compensation in real time without repetition. Accordingly, the super-directional speaker system of the present invention has low delay due to distortion compensation and simplifies the hardware design, making it easy to construct a system capable of highly efficient modulation.

[0086] In other words, the super-directional speaker system according to an embodiment of the present invention uses pre-distortion adaptive filtering to compensate for distortion in the audio input signal in real time, thereby making it possible to pre-distort the audible signal, which is radiated into the air from the ultrasonic transducer and reproduced secondarily, before modulation so that it approaches the original audio input signal. Furthermore, by using a linear FIR filter, the pre-distorted signal is modified within the original bandwidth, and the complexity of the hardware design is reduced. In addition, since the super-directional speaker system of the present invention uses a dynamic modulation method, the sound quality is improved compared to conventional SSB modulation, which uses a non-ideal asymmetric filter, and highly efficient modulation is possible by dynamically changing the modulation index according to the level of the input signal.

[0087] In addition, the super-directional speaker system according to the present invention includes a speaker unit (200) that finally transmits sound waves to the outside, as shown in FIG. 7.

[0088] A super-directional speaker becomes audible to humans in the same way as in the following equation (Berktay's far-field solution).

[0089]

[0090] In the above equation, p(t) represents the sound heard by a person, and env(t) represents the envelope of the signal.

[0091] If env(t) is close to a low frequency (<500Hz), its first derivative p(t) converges to 0. In other words, the lower the frequency of the signal, the closer it is to a direct current (DC) signal, and the derivative of the constant becomes 0, so the signal becomes inaudible.

[0092] To improve this, in an embodiment of the present invention, the speaker unit (200) includes a high frequency (>500Hz) dedicated speaker module (210) and a low frequency (<500Hz) dedicated speaker module (220) to improve the volume of the low frequency band, and preferably, each speaker module is arranged so that the high frequency (>500Hz) dedicated speaker module (210) surrounds the low frequency (<500Hz) dedicated speaker module (220) (Fig. 7).

[0093] At this time, it is desirable to further improve the volume by simultaneously outputting these signals, by separating the signals to be output into bands so that the low-frequency (<500Hz) dedicated speaker module (220) amplifies the volume of the low-frequency signal (preferably ×100 volume) to perform a woofer function, and the high-frequency (>500Hz) dedicated speaker module (210) outputs the high-frequency band signal to perform a tweeter function.

[0094]

[0095] Combinations of each block of the block diagram attached to this specification and each step of the flowchart may be executed by mobile terminal apps or computer program instructions. Since these apps or program instructions may be loaded into the processor of a general-purpose computer, a specialized computer, or other programmable data processing equipment, the instructions executed through the processor of the computer or other programmable data processing equipment create means for performing the functions described in each block of the block diagram or each step of the flowchart. Since these computer program instructions may also be stored in computer-available or computer-readable memory that can be directed toward the computer or other programmable data processing equipment to implement the function in a specific way, the instructions stored in computer-available or computer-readable memory may also produce a manufactured item containing instruction means for performing the function described in each block of the block diagram or each step of the flowchart. Since computer program instructions can be loaded onto a computer or other programmable data processing equipment, instructions that execute a computer or other programmable data processing equipment by performing a series of operation steps on the computer or other programmable data processing equipment to create a process executed by the computer can also provide steps for executing the functions described in each block of the block diagram and each step of the flowchart.

[0096] Additionally, each block or each step may represent a module, segment, or part of code containing one or more executable instructions for executing a specified logical function(s). Also, it should 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 described in succession may actually be performed substantially simultaneously, or the blocks or steps may sometimes be performed in reverse order according to the corresponding function.

[0097] The foregoing description is merely an illustrative explanation of the technical concept of the present invention, and those skilled in the art to which the present invention pertains will be able to make various modifications and variations within the scope of the essential characteristics of the present invention. Accordingly, the embodiments disclosed in the present invention are intended to explain, not limit, the technical concept of the present invention, and the scope of the technical concept of the present invention is not limited by these embodiments. The scope of protection of the present invention shall be interpreted by the claims below, and all technical concepts within an equivalent scope shall be interpreted as being included within the scope of rights of the present invention.

Claims

1. A first envelope calculation unit that calculates the envelope of the currently input audio input signal; A square root calculation unit that generates a square root signal of the first envelope signal by calculating the square root of the first envelope signal calculated by the first envelope calculation unit; A pre-distortion adaptive filter unit that generates a compensation signal by performing distortion compensation by applying an adaptive filter coefficient update term based on the adaptive filter coefficients determined in the previous stage to the currently input audio input signal; A second envelope calculation unit that calculates the envelope of the above compensation signal and generates a second envelope signal; An error calculation unit that generates an error signal by performing a comparison operation between the second envelope signal and the square root signal of the first envelope signal; An adaptive filter coefficient update unit that calculates an adaptive filter coefficient update term and an adaptive filter coefficient from the above error signal; A dynamic signal modulation unit that generates a modulated signal by dynamically modulating the above compensation signal into an ultrasonic band; An ultrasonic transducer model that models an inverse filter corresponding to the frequency characteristics of the ultrasonic transducer and applies it to the modulation signal to generate a filtered signal; An ultrasonic amplifier that amplifies the above filtering signal; and The ultrasonic transducer that converts the amplified signal into an ultrasonic signal is provided, The above dynamic signal modulation unit comprises: an AM signal modulation unit that generates an AM modulation signal of transmitted data; a first sideband signal modulation unit that receives the AM modulation signal and performs a Hilbert operation to obtain a first sideband modulation signal with the carrier removed; and a second sideband signal modulation unit that receives the first sideband modulation signal and adds the carrier to obtain a second sideband modulation signal, characterized by a super-directional speaker system.

2. In Paragraph 1, The above audio input signal is x(t), the above first envelope signal is E(t), and the above previous stage adaptive filter coefficient update term is a m If we say (t), The above compensation signal x(t)' is, ; The second envelope signal E(t)' obtained by AM modulating the above compensation signal x(t)' is E(t)' = 1 + mx(t)'; The above error signal e(t) is, e(t) = E(t)' - E(t) 0.52 ; The above adaptive filter coefficient update term is △a m (t) = - e(t) / a m (t) = -2E(t)' - E(t) 0.5 x(tm); The above adaptive filter coefficients are a m (t+1) = a m (t) + β△a m (t) and, A super-directional speaker system characterized in that, however, m is the modulation index and β is the adaptation coefficient.

3. In Paragraph 2, The above dynamic signal modulation unit A super-directional speaker system characterized by dynamically changing the modulation index according to the input signal level.

4. In Paragraph 1, The above adaptive filter coefficient update unit A super-directional speaker system characterized by applying at least one of the LMS method and the RLS method.

5. In Paragraph 1, The above-mentioned pre-distortion adaptive filter unit A super-directional speaker system characterized by including a linear FIR filter.

6. In Paragraph 1, The above inverse filter is A super-directional speaker system characterized by being pre-calculated using the frequency characteristics of the ultrasonic transducer obtained by modeling the ultrasonic transducer as a specific filter.

7. In claim 6, the specific filter is A super-directional speaker system characterized by being an FIR filter.

8. An adaptive filter calculation unit that calculates current adaptive filter coefficients by comparing the envelope of the current audio input signal with the envelope to which adaptive filter coefficients obtained from the previous audio input signal have been applied; A dynamic signal modulation unit that dynamically modulates an audio input signal to which the above adaptive filter coefficient values ​​are applied; The ultrasonic transducer that converts a modulated signal into ultrasound is provided, The above dynamic signal modulation unit comprises: an AM signal modulation unit that generates an AM modulation signal of transmitted data; a first sideband signal modulation unit that receives the AM modulation signal and performs a Hilbert operation to obtain a first sideband modulation signal with the carrier removed; and a second sideband signal modulation unit that receives the first sideband modulation signal and adds the carrier to obtain a second sideband modulation signal, characterized by a super-directional speaker system.

9. In Paragraph 8, The adaptive filter calculation unit comprises: a first envelope calculation unit that calculates the envelope of a currently input audio input signal; a square root calculation unit that generates a square root signal of the first envelope signal by calculating the square root of the first envelope signal calculated by the first envelope calculation unit; a pre-distortion adaptive filter unit that generates a compensation signal by performing distortion compensation on the currently input audio input signal by applying an adaptive filter coefficient update term according to the adaptive filter coefficients determined in the previous stage; a second envelope calculation unit that generates a second envelope signal by calculating the envelope of the compensation signal; an error calculation unit that generates an error signal by comparing the second envelope signal with the square root signal of the first envelope signal; and an adaptive filter coefficient update unit that calculates an adaptive filter coefficient update term and adaptive filter coefficients from the error signal. The dynamic signal modulation unit dynamically modulates the compensation signal into the ultrasonic band to generate a modulated signal, and A super-directional speaker system characterized by comprising: an ultrasonic transducer model that models an inverse filter corresponding to the frequency characteristics of the ultrasonic transducer and applies it to the modulation signal to generate a filtered signal; an ultrasonic amplifier that amplifies the filtered signal; and the ultrasonic transducer that converts the amplified signal into an ultrasonic signal.

10. A first step of calculating the envelope of the current audio input signal to generate a first envelope signal; A second step of generating an ideal envelope signal of the first envelope signal; A third step of generating a pre-distortion compensated signal by applying adaptive filter coefficients determined by the previous audio input signal; Step 4: Generating the envelope signal of the above compensation signal; A fifth step of generating an error signal by comparing the ideal envelope signal and the envelope signal of the compensation signal; Step 6, calculating an adaptive filter coefficient update term and adaptive filter coefficients from the above error signal; Step 7, generating a modulated signal by dynamically modulating the above compensation signal; Step 8, filtering the above modulated signal with an inverse filter corresponding to an ultrasonic transducer; Step 9, ultrasonically amplifying the filtered signal above; and The above ultrasonic amplified signal is converted into an ultrasonic signal in a tenth step, wherein The modulation in the above 7th step is, (1) a step of obtaining an AM modulation signal of transmitted data; (2) a step of receiving the AM modulation signal and obtaining a first sideband modulation signal consisting only of a first sideband from which the carrier wave has been removed; and (3) a step of receiving the first sideband modulation signal and adding the carrier wave to obtain a second sideband modulation signal, characterized by comprising a super-directional speaker signal processing method.

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