Distortion-free constant beam design method and system based on loudspeaker array

Abstract

The invention discloses a distortion-free constant beam design method and system based on a loudspeaker array, and relates to the technical field of constant beam design. The method comprises: based on the free space sound fields of each loudspeaker array at a preset frequency point, using inverse Fourier transform and a preset window function to establish a first impulse response of a first preset length and a first target response frequency, and using the first impulse response to perform distortion compensation filtering on each loudspeaker array; using the number of points of a preset second length, the number of delayed decimal points, the first number of points, and the delay time, based on a sinc function and a preset window function to establish a second impulse response of a second preset length and a second target response frequency, and using the second impulse response to perform delay compensation filtering on the corresponding loudspeaker array; having the characteristics of a constant beam width, having a more uniform sound reinforcement effect in a specified direction, and the specified direction is electronically adjustable, which increases the flexibility of adjusting the directivity of the loudspeaker.

Description

Technical Field

[0001] The invention relates to the technical field of constant beam design, and more specifically, to a method and system for designing a distortion-free constant beam based on a loudspeaker array. Background Art

[0002] The speaker array beamforming technology uses the principle of wave superposition to send sound signals in a specified direction while attenuating signals in other directions to achieve the purpose of changing the directivity of the speaker. This technology is also known as spatial filtering technology. If the directivity of the speaker is modeled with a baffle piston, the directivity index of the speaker is It can be expressed as follows:

[0003] ,in is a first-order Bessel function of the first kind, is the wave number, is the frequency, It's the speed of sound. is the baffle radius, is the angle between the specified direction and the axis.

[0004] On the other hand, if we assume that each loudspeaker in the array is a point source, the sound field in free space can be expressed as follows:

[0005] ,in is a specified location, is the distance from the specified position to each speaker, is the angular frequency, is the intensity of each speaker, Indicates the power of the speaker, is the relative delay of the loudspeaker, is the number of loudspeakers; the directivity produced by array beamforming is:

[0006] , and the directivity index of the loudspeaker The superposition forms a new directional characteristic. This application proposes a beam design method for the directivity of beamforming, which mainly solves the following deficiencies of traditional solutions:

[0007] 1. Due to the directivity index of the speaker It is relatively fixed, and it is difficult to achieve arbitrary directivity or beam control through speaker design, which is not conducive to the sound reinforcement needs of concentrated sound reinforcement in large scenes, reducing building reflections, reducing building sound equipment costs, and improving sound reinforcement efficiency. The above sound reinforcement needs are widely practical and urgent in sound reinforcement environments such as broadcasting, performances, and conferences;

[0008] 2. From From the expression, we can see that the target response of beamforming is related to frequency. , They are all functions of frequency. Different from beamforming technologies such as wireless networks and radar arrays, the beamforming of speaker arrays needs to be implemented in a wider frequency band, which poses many difficulties for the technical optimization of beamforming. One method of broadband beamforming is to beamform multiple narrowband signals, which greatly increases the amount of beamforming calculations.

[0009] 3. It is difficult for traditional beamforming methods to control the beam width while achieving uniform sound amplification and distortion-free response. Beam optimization methods, such as the interior point method, which is often used in optimization problems, still lack feasible solutions at certain frequencies, and cannot guarantee the similarity of solutions for adjacent frequencies. Therefore, this method is not easy to estimate the bandwidth of beamforming. Inconsistency in solutions over a wide bandwidth may reduce the effect of actual beamforming. This deficiency is further magnified because the audible bandwidth is relatively wide. Summary of the invention

[0010] The purpose of the present invention is to provide a distortion-free constant beam design method and system based on a loudspeaker array, which has the characteristics of a constant beam width, has a relatively uniform sound reinforcement effect in a specified direction, and the specified direction is electronically adjustable, which increases the flexibility of adjusting the directivity of the loudspeaker; at the same time, the frequency response in the specified direction is relatively flat, achieving the purpose of reducing distortion after adjusting the directivity.

[0011] The above technical objectives of the present invention are achieved through the following technical solutions:

[0012] In a first aspect, the present application provides a method for designing a distortion-free constant beam based on a loudspeaker array, comprising the following specific steps:

[0013] According to a plurality of preset equally spaced loudspeaker arrays and arrangement parameters of each loudspeaker array, the spherical radius of the waveform sphere of each loudspeaker array is calculated;

[0014] Using the layout parameters and spherical radius of each speaker array, the intersection of each speaker array and the waveform sphere is calculated, and the relative delay of each speaker array is calculated based on the intersection of the connection line and the actual position of the corresponding speaker array;

[0015] For each speaker array, the speaker angle and the simulation angle in the speaker array arrangement parameters are normalized, and the normalized result is used The gain of the loudspeaker array is calculated by using the first-order Legendre polynomial, and the complex gain of the loudspeaker array is calculated according to the phase difference caused by the gain and the relative time delay;

[0016] According to the actual listening distance and the actual position of each speaker array, and based on the complex gain of each speaker array, a free space sound field of the corresponding speaker array is calculated;

[0017] Based on the free space sound fields of each loudspeaker array at a preset frequency point, an inverse Fourier transform and a preset window function are used to establish a first impulse response of a first preset length and a first target response frequency, and the first impulse response is used to perform distortion compensation filtering on each loudspeaker array.

[0018] Based on the above technical solution, the present invention can also be improved as follows.

[0019] Furthermore, the above method further comprises: for each speaker array,

[0020] The delay point number is calculated by using the relative delay of the speaker array and the preset sampling rate, the decimal place of the delay point number is intercepted as the delay decimal point number, and the delay time of the delay decimal point number is obtained;

[0021] The first number of points required for the second impulse response corresponding to the delayed decimal point is calculated by presetting the second length, and a second impulse response of a second preset length and a second target response frequency is established based on a sinc function and a preset window function using the number of points of the preset second length, the delayed decimal point, the first number of points, and the delay time. The second impulse response is used to perform delay compensation filtering on the corresponding speaker array.

[0022] Furthermore, the spherical radius of the corrugated sphere of each of the above-mentioned speaker arrays is specifically:

[0023] ;

[0024] In the formula, R represents the radius of the sphere, D represents the length of the layout parameters of the corresponding speaker array, represents the emission angle in the arrangement parameters of the corresponding loudspeaker array, Indicates the simulated angle in the arrangement parameters of the corresponding loudspeaker array.

[0025] Furthermore, the above calculation obtains the intersection point of the line connecting each loudspeaker array and the waveform sphere, specifically:

[0026] ; ;

[0027] In the formula, x The horizontal coordinate of the intersection point of the connecting line, y The ordinate of the intersection of the lines. represents the emission angle in the arrangement parameters of the corresponding loudspeaker array, represents the simulated angle in the arrangement parameters of the corresponding loudspeaker array, R represents the spherical radius of the corresponding speaker array, h Indicates the height in the arrangement parameters of the corresponding speaker array.

[0028] Furthermore, the above calculation of the free space sound field of the corresponding speaker array is performed based on the actual listening distance and the actual position of each speaker array and the complex gain of each speaker array, specifically:

[0029] A1, using the actual listening distance and the actual position of the speaker array to calculate the actual distance between the actual listening position and the speaker array, specifically: ; In the formula, Represents a speaker array m Actual distance, r Indicates the actual listening distance. R represents the spherical radius of the corresponding speaker array, h represents the height of the corresponding speaker array layout parameters, represents the emission angle in the arrangement parameters of the corresponding loudspeaker array;

[0030] A2, using the complex gain and actual distance to calculate the free space sound field of the corresponding speaker array, specifically:

[0031] ; In the formula, Represents a speaker array m The complex gain, Indicates the speaker array The gain of each speaker, denote the imaginary unit and the angular frequency of the signal respectively, Represents a speaker array m The relative delay of

[0032] ; In the formula, represents the free space sound field corresponding to the loudspeaker array, Represents a speaker array m The actual distance Represents a speaker array m The complex gain, denote imaginary units and wave numbers, respectively. Indicates the number of units in the speaker array.

[0033] Furthermore, the first impulse response is specifically:

[0034] ;

[0035] ;

[0036] In the formula, represents the first impulse response of the first preset length, is the number of inverse Fourier transform points, represents the center of Fourier transform, Indicates the number of points corresponding to the first preset length, Indicates the first preset length, represents an inverse Fourier transform of the first target response frequency;

[0037] The first target response frequency is specifically:

[0038] ; In the formula, represents the first target response frequency, represents the free space sound field corresponding to the loudspeaker array, Indicates the first preset length, All represent angular frequency, Represents an imaginary unit.

[0039] Furthermore, the second impulse response is specifically:

[0040] ; In the formula, represents the second impulse response, Indicates the number of points corresponding to the second preset length, Indicates the delay decimal point, Indicates the second preset length;

[0041] The second target response frequency is specifically:

[0042] ; In the formula, represents the second target response frequency, denote the imaginary unit and angular frequency respectively, Indicates the delay time of the delayed decimal point.

[0043] In a second aspect, the present application provides a distortion-free constant beam design system based on a loudspeaker array, which is applied to any distortion-free constant beam design method based on a loudspeaker array in the first aspect, comprising:

[0044] A speaker array parameter setting module, used to calculate the spherical radius of the waveform sphere of each speaker array according to a plurality of preset equidistant speaker arrays and the arrangement parameters of each speaker array;

[0045] The delay calculation module is used to calculate the intersection of the lines between each speaker array and the waveform sphere using the arrangement parameters of each speaker array and the spherical radius, and calculate the relative delay of each speaker array according to the intersection of the lines and the actual position of the corresponding speaker array;

[0046] The gain calculation module is used to normalize the speaker angle and the simulation angle in the arrangement parameters of the speaker array for each speaker array, and use the normalized result to calculate the gain of the speaker array. The gain of the loudspeaker array is calculated by using the first-order Legendre polynomial, and the complex gain of the loudspeaker array is calculated according to the phase difference caused by the gain and the relative time delay;

[0047] A pre-simulation module, used for calculating the free space sound field of the corresponding speaker array according to the actual listening distance and the actual position of each speaker array and based on the complex gain of each speaker array;

[0048] The compensation module is used to establish a first impulse response of a first preset length and a first target response frequency based on the free space sound fields of each speaker array at a preset frequency point by using an inverse Fourier transform and a preset window function, and to perform distortion compensation filtering on each speaker array by using the first impulse response.

[0049] Furthermore, the compensation module further includes:

[0050] The first submodule is used to calculate the delay point number by using the relative delay of the speaker array and the preset sampling rate, intercept the decimal place of the delay point number as the delay decimal point number, and obtain the delay time of the delay decimal point number;

[0051] The second submodule is used to calculate the first number of points required for the second impulse response corresponding to the delayed decimal point by presetting the second length, and to establish a second impulse response of a second preset length and a second target response frequency based on a sinc function and a preset window function using the number of points of the preset second length, the delayed decimal point, the first number of points, and the delay time, and to perform delay compensation filtering on the corresponding speaker array using the second impulse response.

[0052] In a third aspect, the present application provides an electronic device comprising a memory, a processor, and a computer program stored in the memory and executable on the processor, wherein the processor implements any one of the methods in the first aspect when executing the computer program.

[0053] In a fourth aspect, the present application provides a non-transitory computer-readable storage medium, which stores computer instructions, and the computer instructions enable a computer to execute any one of the methods in the first aspect.

[0054] Compared with the prior art, the present invention has at least the following beneficial effects:

[0055] In the present application, a first impulse response with a first preset length of points is used to implement distortion compensation filtering of the received signal of each speaker array; and an independent integer delay point delay is performed on each speaker, and a second impulse response with a second preset length of points is used to perform delay compensation filtering on the corresponding speaker array; it has the characteristics of a constant beam width, has a more uniform sound reinforcement effect in a specified direction, and the specified direction is electronically adjustable, which increases the flexibility of adjusting the directivity of the speaker; at the same time, the frequency response in the specified direction is relatively flat, so as to achieve the purpose of reducing distortion after adjusting the directivity.

[0056] In the present application, the strength of each speaker, the relative delay of the speaker, and the angular frequency are independent, which greatly reduces the complexity of beam design and the processing pressure of real-time processing; since similar processing is performed on components of different frequencies, the processed signal is consistent over a wide bandwidth, so that the beamforming effect is insensitive to the actual listening distance, the measurement error of the angle, the measurement error of the distance between the speaker units, the speaker strength, the error of the relative delay of the speaker, and the error of the speed of sound. BRIEF DESCRIPTION OF THE DRAWINGS

[0057] The drawings described herein are used to provide a further understanding of the embodiments of the present invention, constitute a part of this application, and do not constitute a limitation of the embodiments of the present invention. In the drawings:

[0058] Figure 1 A method flow chart of a design method in an embodiment of the present invention;

[0059] Figure 2 A connection diagram of a design system according to an embodiment of the present invention;

[0060] Figure 3 A schematic diagram of the connection of electronic equipment in an embodiment of the present invention;

[0061] Figure 4 is a schematic diagram of beam width and beam response when impulse response is not used to compensate for distortion in an embodiment of the present invention;

[0062] Figure 5 It is a schematic diagram of beam width and beam response when using impulse response to compensate for distortion in an embodiment of the present invention. DETAILED DESCRIPTION

[0063] In order to make the purpose, technical solutions and advantages of the embodiments of the present invention clearer, the technical solutions in the embodiments of the present invention will be clearly and completely described below in conjunction with the drawings in the embodiments of the present invention. Obviously, the described embodiments are part of the embodiments of the present invention, not all of the embodiments. Generally, the components of the embodiments of the present invention described and shown in the drawings here can be arranged and designed in various different configurations.

[0064] Therefore, the following detailed description of the embodiments of the present invention provided in the accompanying drawings is not intended to limit the scope of the invention claimed for protection, but merely represents selected embodiments of the present invention. Based on the embodiments of the present invention, all other embodiments obtained by ordinary technicians in this field without creative work are within the scope of protection of the present invention.

[0065] It should be noted that similar reference numerals and letters denote similar items in the following drawings, and therefore, once an item is defined in one drawing, further definition and explanation thereof is not required in subsequent drawings.

[0066] In the description of the embodiments of the present invention, "plurality" means at least 2.

[0067] Embodiment 1: In order to make the speaker array have the characteristics of constant beam width, have a more uniform sound reinforcement effect in a specified direction, and the specified direction is electronically adjustable, and in order to increase the flexibility of adjusting the directivity of the speaker, and at the same time, the frequency response is relatively flat in the specified direction, and the distortion is reduced after adjusting the directivity, such as Figure 1 As shown, this embodiment provides a distortion-free constant beam design method based on a loudspeaker array, comprising the following specific steps:

[0068] S1, according to a plurality of preset equally spaced loudspeaker arrays and arrangement parameters of each loudspeaker array, the spherical radius of the waveform sphere of each loudspeaker array is calculated.

[0069] In this embodiment, a speaker array with 8 units linearly spaced equidistantly can be used, and the spacing between array units is 0.15m. The emission angle designed by the method in this embodiment can be , the simulation angle can be A constant width beam.

[0070] Optionally, the spherical radius of the corrugated sphere of each of the above speaker arrays is specifically:

[0071] ;

[0072] In the formula, R represents the radius of the sphere, D represents the length of the layout parameters of the corresponding speaker array, represents the emission angle in the arrangement parameters of the corresponding loudspeaker array, Indicates the simulated angle in the arrangement parameters of the corresponding loudspeaker array.

[0073] S2, using the arrangement parameters of each speaker array and the spherical radius, calculate the intersection of the lines between each speaker array and the waveform sphere, and calculate the relative delay of each speaker array according to the intersection of the lines and the actual position of the corresponding speaker array.

[0074] Optionally, the above calculation obtains the intersection point of the line connecting each loudspeaker array and the waveform sphere, specifically:

[0075] ; ;

[0076] In the formula, x The horizontal coordinate of the intersection point of the connecting line, y The ordinate of the intersection of the lines. represents the emission angle in the arrangement parameters of the corresponding loudspeaker array, represents the simulated angle in the arrangement parameters of the corresponding loudspeaker array, R represents the spherical radius of the corresponding speaker array, h Indicates the height in the arrangement parameters of the corresponding speaker array.

[0077] Specifically, a coordinate system needs to be established first. Each speaker array can be used as the coordinate origin, the horizontal direction as the x-axis, and the vertical direction as the y-axis. The distance can then be calculated using the horizontal and vertical coordinates of the intersection of the connecting lines and the coordinates of the actual speaker position, and then divided by the speed of sound to obtain the relative delay of each speaker array.

[0078] S3, for each speaker array, normalize the speaker angle and the simulation angle in the arrangement parameters of the speaker array, and use the normalized result to calculate the angle of the speaker. The gain of the loudspeaker array is calculated using the first-order Legendre polynomial, and the complex gain of the loudspeaker array is calculated based on the gain and the phase difference caused by the relative time delay.

[0079] Among them, the emission angle of each speaker For simulation angle Normalized, that is , reuse The gain of each loudspeaker is calculated using the Legendre polynomial of order , that is, .

[0080] S4, calculating the free space sound field of the corresponding speaker array according to the actual listening distance and the actual position of each speaker array and based on the complex gain of each speaker array.

[0081] Optionally, the above calculation of the free space sound field of the corresponding speaker array based on the actual listening distance and the actual position of each speaker array and the complex gain of each speaker array is specifically as follows:

[0082] A1, using the actual listening distance and the actual position of the speaker array to calculate the actual distance between the actual listening position and the speaker array, specifically: ; In the formula, Represents a speaker array m Actual distance, r Indicates the actual listening distance. R represents the spherical radius of the corresponding speaker array, h represents the height of the corresponding speaker array layout parameters, It represents the emission angle in the arrangement parameters of the corresponding speaker array.

[0083] A2, using the complex gain and actual distance to calculate the free space sound field of the corresponding speaker array, specifically:

[0084] ; In the formula, Represents a speaker array m The complex gain, Indicates the speaker array The gain of each speaker, denote the imaginary unit and the angular frequency of the signal respectively, Represents a speaker array m relative delay.

[0085] ; In the formula, represents the free space sound field corresponding to the loudspeaker array, Represents a speaker array m The actual distance Represents a speaker array m The complex gain, denote imaginary units and wave numbers, respectively. Indicates the number of units in the speaker array.

[0086] S5, based on the free space sound fields of each speaker array at the preset frequency point, using inverse Fourier transform and a preset window function to establish a first impulse response of a first preset length and a first target response frequency, and using the first impulse response to perform distortion compensation filtering on each speaker array.

[0087] Optionally, the first impulse response is specifically:

[0088] ;

[0089] ;

[0090] In the formula, represents the first impulse response of the first preset length, is the number of inverse Fourier transform points, represents the center of Fourier transform, Indicates the number of points corresponding to the first preset length, Indicates the first preset length, Represents the inverse Fourier transform of the first target response frequency.

[0091] Furthermore, the first target response frequency is specifically:

[0092] ; In the formula, represents the first target response frequency, represents the free space sound field corresponding to the loudspeaker array, Indicates the first preset length, All represent angular frequency, Represents an imaginary unit.

[0093] Specifically, after the initialization of the beam design, the received signals of each speaker array can be processed in real time. In principle, point-by-point processing can be used to adapt to different sizes of output and input buffer sizes. Specifically, the following steps are taken: 1. Before Cache points are moved back, the received data points are stored, and the impulse response is convolved with the buffer; 2. The length is The effective utilization space of the cache is Add the integer delay point minus 1, shift the effective utilization point minus 1 data point back, store the result of the previous convolution, and then perform the convolution on the integer delay point. Each data point is The length is Delayed convolution.

[0094] Optionally, the above method further includes: for each speaker array,

[0095] S6, using the relative delay of the speaker array and the preset sampling rate to calculate the delay point number, intercepting the decimal place of the delay point number as the delay decimal point number, and obtaining the delay time of the delay decimal point number.

[0096] S7, calculating the first number of points required for the second impulse response corresponding to the delayed decimal point by presetting the second length, using the number of points of the preset second length, the delayed decimal point, the first number of points, and the delay time, based on the sinc function and the preset window function, establish a second impulse response of the second preset length and the second target response frequency, and use the second impulse response to perform delay compensation filtering on the corresponding speaker array.

[0097] The window function used in this embodiment may be a Hanning window, that is: .

[0098] Optionally, the second impulse response is specifically:

[0099] ; In the formula, represents the second impulse response, Indicates the number of points corresponding to the second preset length, Indicates the delay decimal point, Indicates the second preset length.

[0100] Furthermore, the second target response frequency is specifically:

[0101] ; In the formula, represents the second target response frequency, denote the imaginary unit and angular frequency respectively, Indicates the delay time of the delayed decimal point.

[0102] Specifically, the appropriate size of the delay buffer of the 8-speaker array can be determined by the following method: 1. Initialize the maximum delay to 0; 2. Determine the emission angle , the simulation angle is , spherical radius , calculate the intersection point of the speaker and the sphere , and the relative delay is calculated by dividing the distance between the intersection point on the sphere and the actual speaker position by the speed of sound ; 3. Multiply the relative delay by the sampling rate to get the delay point number. If the delay point number is greater than the maximum delay, the maximum delay is the current delay point number; 4. At the output angle arrive Range, simulation angle arrive Repeat steps 2 and 3 within the range; taking this example, at a sampling rate of 48kHz, an 8-unit loudspeaker array with an array unit spacing of 0.15m, the appropriate cache size is , it is not necessary to use the entire cache space, and the cache space size can be further optimized according to the specific exit angle and simulation angle.

[0103] Furthermore, the following method is used to verify the effectiveness of the distortion-free constant beamforming based on the speaker array in this embodiment:

[0104] 1. Calculate the length The impulse response Actual frequency response:

[0105] ;

[0106] 2. Utilize the gain of each speaker array and the phase difference caused by relative delay Multiply the impulse response Actual frequency response Synthesize the complex gain of each speaker ;

[0107] 3. Determine the listening distance , 2048 frequency points can be sampled and calculated , and normalize the obtained response: .

[0108] Take the beam width at -6dB as the reference of the beam width, and obtain the beam width and beam response curve as follows: Figure 4 and Figure 5 As shown; among them, from Figure 4 As can be seen from the figure, when impulse response is not used to compensate for distortion, although the beam width can be maintained within a certain frequency range, the sound intensity is uniform within this range, achieving the effect of uniform sound reinforcement, but the axial frequency response begins to show obvious attenuation in the mid-high frequency, and the sound near the axis has already shown obvious distortion; Figure 5 It can be seen that when using impulse response to compensate for distortion, the beam width is not affected, but the axial frequency response is still flat over a large frequency range, which significantly reduces distortion and improves the sound reinforcement quality. At the same time, the frequency response is also improved in the direction near the axial direction, which shows that the method used maintains the effect of uniform sound reinforcement and improves distortion to achieve the desired effect.

[0109] Embodiment 2: The present application provides a distortion-free constant beam design system based on a loudspeaker array, which is applied to any distortion-free constant beam design method based on a loudspeaker array in Embodiment 1, such as Figure 2 As shown, including:

[0110] The loudspeaker array parameter setting module is used to calculate the spherical radius of the wave sphere of each loudspeaker array according to a plurality of preset equidistant loudspeaker arrays and the arrangement parameters of each loudspeaker array.

[0111] The delay calculation module is used to calculate the intersection of the lines between each speaker array and the waveform sphere using the layout parameters of each speaker array and the spherical radius, and calculate the relative delay of each speaker array based on the intersection of the lines and the actual position of the corresponding speaker array.

[0112] The gain calculation module is used to normalize the speaker angle and the simulation angle in the arrangement parameters of the speaker array for each speaker array, and use the normalized result to calculate the gain of the speaker array. The gain of the loudspeaker array is calculated using the first-order Legendre polynomial, and the complex gain of the loudspeaker array is calculated based on the gain and the phase difference caused by the relative time delay.

[0113] The pre-simulation module is used to calculate the free space sound field of the corresponding speaker array according to the actual listening distance and the actual position of each speaker array and based on the complex gain of each speaker array.

[0114] The compensation module is used to establish a first impulse response of a first preset length and a first target response frequency based on the free space sound fields of each speaker array at a preset frequency point by using an inverse Fourier transform and a preset window function, and to perform distortion compensation filtering on each speaker array by using the first impulse response.

[0115] Optionally, the compensation module may further include:

[0116] The first submodule is used to calculate the delay points by using the relative delay of the speaker array and the preset sampling rate, intercept the decimal places of the delay points as the delay decimal points, and obtain the delay time of the delay decimal points.

[0117] The second submodule is used to calculate the first number of points required for the second impulse response corresponding to the delayed decimal point by presetting the second length, and to establish a second impulse response of a second preset length and a second target response frequency based on a sinc function and a preset window function using the number of points of the preset second length, the delayed decimal point, the first number of points, and the delay time, and to perform delay compensation filtering on the corresponding speaker array using the second impulse response.

[0118] Embodiment 3: This embodiment of the present application provides an electronic device, such as Figure 3 As shown, it includes a memory, a processor, and a computer program stored in the memory and executable on the processor. When the processor executes the computer program, any method in Embodiment 1 is implemented.

[0119] Embodiment 4: The embodiment of the present application provides a non-transitory computer-readable storage medium, which stores computer instructions, and the computer instructions enable a computer to execute any method in Embodiment 1.

[0120] The specific implementation methods described above further illustrate the objectives, technical solutions and beneficial effects of the present invention in detail. It should be understood that the above description is only a specific implementation method of the present invention and is not intended to limit the scope of protection of the present invention. Any modifications, equivalent substitutions, improvements, etc. made within the spirit and principles of the present invention should be included in the scope of protection of the present invention.

Claims

1. A distortion-free constant beam design method based on a loudspeaker array, characterized in that: The specific steps include: According to a plurality of preset equally spaced loudspeaker arrays and arrangement parameters of each loudspeaker array, the spherical radius of the waveform sphere of each loudspeaker array is calculated; Using the arrangement parameters of each loudspeaker array and the radius of the sphere, the intersection of the lines between each loudspeaker array and the waveform sphere is calculated, and the relative delay of each loudspeaker array is calculated according to the intersection of the lines and the actual position of the corresponding loudspeaker array; For each of the speaker arrays, the speaker angle and the simulation angle in the arrangement parameters of the speaker array are normalized, and based on the result of the normalization, the speaker angle and the simulation angle are normalized. The gain of the loudspeaker array is calculated by using the first-order Legendre polynomial, and the complex gain of the loudspeaker array is calculated according to the phase difference caused by the gain and the relative time delay; According to the actual listening distance and the actual position of each speaker array, and based on the complex gain of each speaker array, a free space sound field of the corresponding speaker array is calculated; Based on the free space sound fields of each loudspeaker array at a preset frequency point, an inverse Fourier transform and a preset window function are used to establish a first impulse response of a first preset length and a first target response frequency, and the first impulse response is used to perform distortion compensation filtering on each loudspeaker array.

2. The method for designing a distortion-free constant beam based on a loudspeaker array according to claim 1, characterized in that: The method further comprises: for each of the speaker arrays, The delay point number is calculated by using the relative delay of the speaker array and the preset sampling rate, the decimal place of the delay point number is intercepted as the delay decimal point number, and the delay time of the delay decimal point number is obtained; A first number of points required for a second impulse response corresponding to a delayed decimal point is calculated by presetting a second length, and a second impulse response of a second preset length and a second target response frequency is established based on a sinc function and a preset window function using the number of points of the preset second length, the delayed decimal point, the first number of points, and the delay time, and the second impulse response is used to perform delay compensation filtering on the corresponding speaker array.

3. The method for designing a distortion-free constant beam based on a loudspeaker array according to claim 1, characterized in that: The spherical radius of the corrugated sphere of each speaker array is specifically: ； In the formula, R represents the radius of the sphere, D represents the length of the layout parameters of the corresponding speaker array, represents the emission angle in the arrangement parameters of the corresponding loudspeaker array, Indicates the simulated angle in the arrangement parameters of the corresponding loudspeaker array.

4. The method for designing a distortion-free constant beam based on a loudspeaker array according to claim 1, characterized in that: The calculation obtains the intersection of the line connecting each loudspeaker array and the waveform sphere, specifically: ； ； In the formula, x The horizontal coordinate of the intersection point of the connecting line, y The ordinate of the intersection of the connecting lines. represents the emission angle in the arrangement parameters of the corresponding loudspeaker array, represents the simulated angle in the arrangement parameters of the corresponding loudspeaker array, R represents the spherical radius of the corresponding speaker array, h Indicates the height in the arrangement parameters of the corresponding speaker array.

5. The method for designing a distortion-free constant beam based on a loudspeaker array according to claim 1, characterized in that: According to the actual listening distance and the actual position of each speaker array, the free space sound field of the corresponding speaker array is calculated based on the complex gain of each speaker array, specifically: A1, using the actual listening distance and the actual position of the speaker array to calculate the actual distance between the actual listening position and the speaker array, specifically: ; In the formula, Represents a speaker array m Actual distance, r Indicates the actual listening distance. R represents the spherical radius of the corresponding speaker array, h represents the height of the corresponding speaker array layout parameters, represents the emission angle in the arrangement parameters of the corresponding loudspeaker array; A2, using the complex gain and actual distance to calculate the free space sound field of the corresponding speaker array, specifically: ; In the formula, Represents a speaker array m The complex gain, Indicates the speaker array The gain of each speaker, denote the imaginary unit and the angular frequency of the signal respectively, Represents a speaker array m The relative delay of ; In the formula, represents the free space sound field corresponding to the loudspeaker array, Represents a speaker array m The actual distance Represents a speaker array m The complex gain, denote imaginary units and wave numbers, respectively. Indicates the number of units in the speaker array.

6. The method for designing a distortion-free constant beam based on a loudspeaker array according to claim 1, characterized in that: The first impulse response is specifically: ； ； In the formula, represents the first impulse response of the first preset length, is the number of inverse Fourier transform points, represents the center of Fourier transform, Indicates the number of points corresponding to the first preset length, Indicates the first preset length, represents an inverse Fourier transform of the first target response frequency; The first target response frequency is specifically: ; In the formula, represents the first target response frequency, represents the free space sound field corresponding to the loudspeaker array, Indicates the first preset length, All represent angular frequency, Represents an imaginary unit.

7. The method for designing a distortion-free constant beam based on a loudspeaker array according to claim 2, characterized in that: The second impulse response is specifically: ; In the formula, represents the second impulse response, Indicates the number of points corresponding to the second preset length, Indicates the delay decimal point, Indicates the second preset length; The second target response frequency is specifically: ; In the formula, represents the second target response frequency, denote the imaginary unit and angular frequency respectively, Indicates the delay time of the delayed decimal point.

8. A distortion-free constant beam design system based on a loudspeaker array, applied to a distortion-free constant beam design method based on a loudspeaker array according to any one of claims 1 to 7, characterized in that: include: A speaker array parameter setting module, used to calculate the spherical radius of the waveform sphere of each speaker array according to a plurality of preset equidistant speaker arrays and the arrangement parameters of each speaker array; A delay calculation module, used to calculate the intersection of the lines between each speaker array and the waveform sphere using the arrangement parameters of each speaker array and the spherical radius, and calculate the relative delay of each speaker array according to the intersection of the lines and the actual position of the corresponding speaker array; The gain calculation module is used to normalize the speaker angle and the simulation angle in the arrangement parameters of the speaker array for each of the speaker arrays, and use the normalized result to calculate the gain. The gain of the loudspeaker array is calculated by using the first-order Legendre polynomial, and the complex gain of the loudspeaker array is calculated according to the phase difference caused by the gain and the relative time delay; A pre-simulation module, used for calculating the free space sound field of the corresponding speaker array according to the actual listening distance and the actual position of each speaker array and based on the complex gain of each speaker array; A compensation module is used to establish a first impulse response of a first preset length and a first target response frequency based on the free space sound field of each speaker array at a preset frequency point by using an inverse Fourier transform and a preset window function, and to perform distortion compensation filtering on each speaker array by using the first impulse response.

9. The distortion-free constant beam design system based on a loudspeaker array according to claim 8, characterized in that: The compensation module also includes: The first submodule is used to calculate the delay point number by using the relative delay of the speaker array and the preset sampling rate, intercept the decimal place of the delay point number as the delay decimal point number, and obtain the delay time of the delay decimal point number; The second submodule is used to calculate the first number of points required for the second impulse response corresponding to the delayed decimal point by presetting the second length, and to establish a second impulse response of a second preset length and a second target response frequency based on a sinc function and a preset window function by using the number of points of the preset second length, the delayed decimal point, the first number of points, and the delay time, and to perform delay compensation filtering on the corresponding speaker array by using the second impulse response.

10. An electronic device, characterized in that: The method comprises a memory, a processor and a computer program stored in the memory and executable on the processor, wherein the method according to any one of claims 1 to 7 is implemented when the processor executes the computer program.

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