CN117098045A - Array loudspeaker system and implementation method - Google Patents
Array loudspeaker system and implementation method Download PDFInfo
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Classifications
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R9/00—Transducers of moving-coil, moving-strip, or moving-wire type
- H04R9/06—Loudspeakers
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R9/00—Transducers of moving-coil, moving-strip, or moving-wire type
- H04R9/02—Details
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R2400/00—Loudspeakers
- H04R2400/11—Aspects regarding the frame of loudspeaker transducers
Abstract
The invention provides an array loudspeaker system and an implementation method, which relate to the technical field of loudspeakers, wherein the array loudspeaker system comprises: the input end analog-to-digital conversion module is used for carrying out analog or digital conversion on an input audio signal source to obtain a digital signal; the signal matrix module is used for dividing the digital signals to obtain multiple paths of signals; and the output end processing module is used for carrying out time domain and frequency domain joint processing on the multipath signals to realize the control of the wave field distribution of the loudspeaker array. The invention can improve the array loudspeaker system with the wave field evenly distributed in the specific direction angle and azimuth angle, optimize the multi-directional sound propagation, and lead the sound to have higher concentration and uniformity in the specific direction.
Description
Technical Field
The invention relates to the technical field of speakers, in particular to an array speaker system and an implementation method.
Background
The types of loudspeakers currently on the market mainly include the following:
line array speaker (Line Array Speakers): the line array loudspeaker is a line array, arc or J-shaped array loudspeaker array, which aims to realize long-distance sound transmission. However, wire array speakers can present challenges in achieving uniform sound field distribution, as sound may be affected differently at different height audience locations, as well as reflection and diffraction problems.
An adjustable directivity array speaker: the deflection angle of the wave beam can be changed by the loudspeaker array, and the sound expansion effect can be dynamically adjusted along with the change of the sound expansion requirement. The beam control method of the loudspeaker array is simpler, the sound pressure level of the generated sound field is concentrated at the center of the main lobe of the sound beam, so that the sound field is unevenly distributed, and different phase interference can be generated along with the deflection of angles.
Point array speaker (Point Source Array Speakers): the point array speaker places a plurality of speakers in one area to achieve uniform distribution of sound. However, phase matching and reflection problems may affect the quality and distribution of sound.
Omni speaker (Omni-Directional Speakers): omni-directional speakers attempt to provide the same sound distribution in all directions. However, omni-directional speakers may not achieve truly uniform sound field distribution due to reflection, refraction, diffraction, etc. of sound.
Stereo/surround sound system (Stereo/Surround Sound Systems): these systems typically place speakers in multiple directions to create stereo or surround sound effects in both the horizontal and vertical directions. However, refraction, reflection and superposition of sound may lead to uneven distribution of sound between listeners at different locations.
The above speakers have the following main disadvantages:
sound field inhomogeneity: it may result in non-uniform sound fields, i.e. stronger sound in certain areas and weaker sound in other areas, or the superposition and interference of sound phenomena that affect the hearing experience of the listener at different locations.
Sound diffraction and refraction: in a loudspeaker array, diffraction and refraction phenomena of sound occur, resulting in sound propagating in different directions. Wave-field-free uniform distribution techniques may not be able to effectively control these phenomena, affecting accurate communication and localization of sound.
Phase matching is difficult: in the technology of uniform distribution of no wave field, phase matching between speakers is very critical, but due to environmental changes, equipment differences, etc., achieving accurate phase matching may become complicated, affecting sound quality and sound field performance.
Reflection and reverberation problems: in a practical environment, sound may be reflected by walls, ceilings, etc., and reverberation occurs, affecting the clarity and positioning of the sound. Wave-free field uniform distribution techniques may have difficulty coping with these reflection and reverberation problems.
Energy loss: to achieve a uniform wave field distribution, the speakers may need to operate at lower power to reduce mutual interference. This may result in a lower volume, especially when a larger area needs to be covered.
Disclosure of Invention
The invention aims to provide an array loudspeaker system and an implementation method thereof, which can improve the uniform distribution of wave fields in specific direction angles and azimuth angles, optimize the multi-directional sound transmission and ensure that the sound has higher concentration and uniformity in the specific direction.
In order to solve the technical problems, the technical scheme of the invention is as follows:
in a first aspect, an array speaker system includes:
the input end analog-to-digital conversion module is used for carrying out analog or digital conversion on an input audio signal source to obtain a digital signal;
the signal matrix module is used for dividing the digital signals to obtain multiple paths of signals;
and the output end processing module is used for carrying out time domain and frequency domain joint processing on the multipath signals to realize the control of the wave field distribution of the loudspeaker array.
Further, the output end processing module includes:
the time domain processing module is used for setting delay time of the multipath signals;
the frequency domain processing module is used for processing each path of signal after delay setting through the high-low pass filter module;
the FIR filter module is used for controlling the channel phase and the amplitude of the signals processed by the high-low pass filter module;
the multichannel power amplifier module is used for amplifying signals after channel phase and amplitude control;
and the array loudspeaker is used for converting the amplified and output electric signals into sound waves.
In a second aspect, a method for implementing an array speaker system, the method comprising:
determining the number, spacing and frequency parameters of the loudspeakers;
calculating a position vector from each speaker to a reference point based on the array structure;
calculating a unit direction vector of the wave beam according to the requirement;
according to the position vector and the beam direction, calculating the time delay parameter of each loudspeaker by applying a beam forming formula, and according to the operation frequency, calculating the wavelength of the sound wave;
calculating a normalized distance parameter required by the Bessel function based on the position and the wavelength of the loudspeaker, substituting the position of the loudspeaker into the Bessel function, and calculating the amplitude weight and the phase weight of each loudspeaker;
the time domain processing module controls the delay of each loudspeaker according to the time delay parameter, the frequency domain processing module controls the amplitude and the phase of each loudspeaker according to the weight, amplifies the processed driving signals, and the loudspeaker unit converts the amplified electric signals into sound waves.
Further, determining the number of speakers, the spacing, and the frequency parameters includes:
an array speaker system is set, which is arranged in an array and uniformly spaced, and consists of N identical speaker units, in a specific direction theta 1 and azimuth angleForming a beam thereon;
the radiation frequency of the loudspeaker is f, the sound velocity is c, the cell spacing is d, and a spherical wave model is used,representing the azimuth angle of the array speaker system, in the spherical coordinate system, the polar coordinates for the sound fieldWherein θ2 represents the angle between the direction of propagation of the sound wave and the vertical direction; />Representing the angle between the direction of propagation of the sound wave and the reference direction.
Further, based on the array structure, calculating a position vector of each speaker to a reference point includes:
calculating d (i) a position vector from the ith element to a reference point of the array, the reference point being provided with a first speaker unit of the array, the array being provided in a cartesian three-dimensional coordinate system, each position being represented by three coordinate values: x, y and z, the position of the ith speaker is (x (i), y (i), z (i)), the geometric center of the array is selected as a reference point, the coordinates are (x (ref), y (ref), z (ref)), the position vector between the two points is represented by the difference of Cartesian coordinates, and the calculation formula of d (i) is:
further, according to the requirement, calculating the unit direction vector of the beam includes:
calculating a unit direction vector s, s representing a direction from the reference point to the arrival of the sound source, the direction being defined by the azimuth angle in the spherical coordinate systemAnd a polar angle θ3, for a given azimuth and polar angle, the unit direction vector calculation formula is:
wherein,is the azimuth angle, defined as the angle from the x-axis to the direction vector projected onto the xy-plane, the range is (0, 2pi), θ3 is the elevation angle, defined as the angle fromThe angle of the z-axis to the direction vector ranges from (0, pi).
Further, according to the position vector and the beam direction, a beam forming formula is applied to calculate a delay parameter of each speaker, and according to the operation frequency, a wavelength of the sound wave is calculated, including:
and calculating the time delay of the array beam pointing angle theta 4, wherein a time delay beam forming formula is as follows:
T(i)＝d(i)×s/c，
wherein, the time delay of the ith element of T (i), d (i) is a vector from the ith element to a reference point of the array, s is a beam unit direction vector, c is sound velocity, and the relation between the wavelength lambda of the sound wave and the frequency f and the sound velocity c is: λ=c/f.
Further, calculating normalized distance parameters required for the Bessel function based on the speaker position and wavelength, including:
parameters required for the bessel function are calculated, the distance norm_d is normalized, and the calculation formula is norm_d=d/λ, where d is the cell spacing and λ is the wavelength.
Further, substituting the speaker position into the bessel function, and calculating the amplitude weight and the phase weight of each speaker includes:
calculating the position x of each loudspeaker;
setting the order value n of different frequencies, substituting the order value n into a Bessel function, wherein the formula is as follows:
wherein t is an integral variable;
calculating the amplitude weight A of each loudspeaker unit and setting the real value J of the Bessel function n (x) Take the absolute value, i.e., a= |jn (x) |.
Further, the calculation formula of the position x of each speaker is:
n represents the total number of samples, +.>Representing the normalized distance value.
The scheme of the invention at least comprises the following beneficial effects:
according to the scheme, the propagation direction of sound waves can be precisely controlled, beam forming is realized, sound is directed to a specific direction, the direction of sound wave beams can be continuously changed by adjusting time delay parameters, rapid scanning is realized, bessel functions are applied to calculate amplitude and phase weights, the sidelobe level of the beams can be optimized, the directionality of the beams is improved, the sound beam focusing power is increased, the beam forming can be realized through software digital control without mechanical operation, the control is flexible, very narrow beam width can be generated, long-distance and high-directionality sound wave propagation is realized, the system structure is modularized, a large-scale array can be conveniently assembled, more precise beam control is realized, optimal beam forming parameters can be calculated according to different requirements and environments, self-adaptive beam control is realized, the sound field directivity can be comprehensively optimized through joint processing of frequency domains and time domains, the system performance is improved, and the method realizes precise, high-efficiency, controllable and optimizable beam forming can be widely applied to the fields of sonar, acoustic communication and the like.
Drawings
Fig. 1 is a schematic diagram of an array speaker system according to an embodiment of the present invention.
Fig. 2 is a flow chart of an implementation method of an array speaker system according to an embodiment of the present invention.
Detailed Description
Exemplary embodiments of the present disclosure will be described more closely below with reference to the accompanying drawings. While exemplary embodiments of the present disclosure are shown in the drawings, it should be understood that the present disclosure may be embodied in various forms and should not be limited to the embodiments set forth herein. Rather, these embodiments are provided so that this disclosure will be thorough and complete, and will fully convey the scope of the disclosure to those skilled in the art.
As shown in fig. 1, an embodiment of the present invention proposes an array speaker system including:
the input end analog-to-digital conversion module 1 is used for carrying out analog or digital conversion on an input audio signal source to obtain a digital signal;
the signal matrix module 2 is used for dividing the digital signals to obtain multiple paths of signals;
the output end processing module is used for carrying out time domain and frequency domain combined processing on the multipath signals to realize control on wave field distribution of the loudspeaker array; the output end processing module comprises:
the time domain processing module 3 is used for setting delay time of the multipath signals;
the frequency domain processing module is used for processing each path of signal after delay setting through the high-low pass filter module 4;
the FIR filter module 5 is used for controlling the channel phase and amplitude of the signal processed by the high-low pass filter module 4;
a multi-channel power amplifier module 6 for amplifying the signals after channel phase and amplitude control;
and an array speaker 7 for converting the amplified output electric signal into an acoustic wave.
In the embodiment of the invention, an input end analog-to-digital conversion module 1 converts an analog audio signal into a digital signal; the signal matrix module 2 divides the digital signals into multiple paths corresponding to each subsequent speaker channel; a time domain processing module 3 for applying a delay to each channel signal for time domain control of beam forming; the high-low pass filter module 4 performs frequency band screening on the signals to improve the stability of the system; the FIR filter module 5 is used for precisely controlling the amplitude and the phase of each channel and controlling the frequency domain of beam forming; a multi-channel power amplifier module 6 that amplifies the signal to a level sufficient to drive the speaker; the array speaker 7 converts the electric signal into sound and outputs the sound, and forms a beam-formed speaker array.
A method of implementing an array speaker system, the method comprising:
step 11, determining the number, the spacing and the frequency parameters of the loudspeakers;
step 12, calculating the position vector from each loudspeaker to the reference point based on the array structure;
step 13, calculating a unit direction vector of the wave beam according to the requirement;
step 14, calculating the time delay parameter of each loudspeaker by applying a beam forming formula according to the position vector and the beam direction, and calculating the wavelength of the sound wave according to the operation frequency;
step 15, calculating a normalized distance parameter required by the Bessel function based on the position and the wavelength of the loudspeaker, substituting the position of the loudspeaker into the Bessel function, and calculating the amplitude weight and the phase weight of each loudspeaker;
step 16, in the time domain processing module, the delay of each speaker is controlled according to the delay parameter, in the frequency domain processing module, the amplitude and the phase of each speaker are controlled according to the weight, the processed driving signals are amplified, and the speaker unit converts the amplified electric signals into sound waves.
In the embodiment of the invention, the propagation direction of sound waves can be precisely controlled, the beam forming is realized, the sound is directed to a specific direction, the direction of the sound wave beam can be continuously changed by adjusting time delay parameters, the rapid scanning is realized, the Bessel function is applied to calculate the amplitude and phase weight, the sidelobe level of the beam can be optimized, the directionality of the beam is improved, the sound beam focusing power is increased, the beam forming can be realized through software digital control without mechanical operation, the control is flexible, the very narrow beam width can be generated, the long-distance and high-directionality sound wave propagation is realized, the system structure is modularized, the large-scale array can be conveniently assembled, the more precise beam control is realized, the optimal beam forming parameters can be calculated according to different requirements and environments, the self-adaptive beam control is realized, the sound field directivity can be comprehensively optimized through the joint processing of the frequency domain and the time domain, the system performance is improved, the method realizes the precise, the high-efficiency, the controllable and the optimized beam forming can be widely applied to the fields of sonar, the sound communication and the like.
In a preferred embodiment of the invention, determining the number of loudspeakers, spacing, frequency parameters comprises:
setting array-arranged and uniformly-spaced array loudspeaker systems consisting of N identical loudspeakersA unit assembly in a specific direction theta 1 and azimuth angleForming a beam thereon;
the radiation frequency of the loudspeaker is f, the sound velocity is c, the cell spacing is d, and a spherical wave model is used,representing the azimuth angle of the array loudspeaker system, wherein in a spherical coordinate system, the sound field is represented by polar coordinates, and theta 2 represents the included angle between the propagation direction of sound waves and the vertical direction; />Representing the angle between the direction of propagation of the sound wave and the reference direction.
In the embodiment of the invention, the structural layout of the array can be explicitly described; the number N of the loudspeakers is set, so that the array length can be controlled, and the beam width is influenced; the distance d is set, so that diffraction errors and aliasing effects can be avoided; setting a frequency f to ensure that the loudspeaker works in an effective frequency band of the loudspeaker; describing a coordinate system and a direction angle, the beam direction can be accurately represented; the spherical wave model is applied to more accurately describe the sound field distribution; the parameters provide necessary structural information for calculating the beam forming parameters such as time delay, phase and the like, are favorable for establishing a mathematical model subsequently and carrying out beam forming and optimization, can be adjusted to change the beam shape and control effect, provide a foundation for designing beam forming systems with different requirements, are favorable for understanding and analyzing the principle of beam forming, guide parameter selection and determine reasonable parameters, and are preconditions for realizing beam control.
In a preferred embodiment of the present invention, calculating a position vector of each speaker to a reference point based on an array structure includes:
calculating d (i) a position vector from the ith element to a reference point of the array, the reference point being provided with a first speaker unit of the array, the array being provided in a cartesian three-dimensional coordinate system, each position being represented by three coordinate values: x, y and z, the position of the ith speaker is (x (i), y (i), z (i)), the geometric center of the array is selected as a reference point, the coordinates are (x (ref), y (ref), z (ref)), the position vector between the two points is represented by the difference of Cartesian coordinates, and the calculation formula of d (i) is:
in the embodiment of the invention, the accurate position coordinates of each loudspeaker in the array can be definitely obtained, position information is provided for calculating time delay and phase parameters of each loudspeaker in beam forming, the position vectors can be expressed in different coordinate systems through coordinate transformation, the method is favorable for analyzing the distance and direction from each loudspeaker to a sound field designated position, the method can be applied to loudspeaker arrays with various structures, the beam shape can be optimized through adjusting the array structure, a foundation is laid for realizing an adaptive beam forming algorithm, the propagation characteristics of sound waves can be analyzed, the sound pressure distribution can be calculated, the position vectors can be applied to other beam forming models through coordinate transformation, the position information foundation is provided for realizing beam scanning, and the method is favorable for understanding and analyzing the mathematical principle of beam forming.
In a preferred embodiment of the present invention, calculating the unit direction vector of the beam according to the requirement includes:
calculating a unit direction vector s, s representing a direction from the reference point to the arrival of the sound source, the direction being defined by the azimuth angle in the spherical coordinate systemAnd a polar angle θ3, for a given azimuth and polar angle, the unit direction vector calculation formula is:
wherein,is azimuth angleThe range is (0, 2pi), the elevation angle, θ3, the angle from the z-axis to the direction vector, and the range is (0, pi).
In the embodiment of the invention, by setting the azimuth angleAnd the polar angle theta 3 parameter can accurately control the beam direction, can convert the directions of different coordinate systems into uniform unit vectors, is favorable for calculating the driving delay of a loudspeaker in beam forming, can quickly and continuously change the vectors to realize beam scanning, and can be used together with the position vectors for direction propagation modeling.
In a preferred embodiment of the present invention, the time delay parameter of each speaker is calculated by applying a beamforming formula according to the position vector and the beam direction, and the wavelength of the sound wave is calculated according to the operation frequency, including:
and calculating the time delay of the array beam pointing angle theta 4, wherein a time delay beam forming formula is as follows:
T(i)＝d(i)×s/c，
wherein, the time delay of the ith element of T (i), d (i) is a vector from the ith element to a reference point of the array, s is a beam unit direction vector, c is sound velocity, and the relation between the wavelength lambda of the sound wave and the frequency f and the sound velocity c is: λ=c/f.
In a preferred embodiment of the invention, calculating the normalized distance parameters required for the Bessel function based on the speaker position and wavelength comprises:
parameters required for the bessel function are calculated, the distance norm_d is normalized, and the calculation formula is norm_d=d/λ, where d is the cell spacing and λ is the wavelength.
In a preferred embodiment of the present invention, substituting the speaker position into the bessel function, the amplitude weight and the phase weight of each speaker are calculated, including:
calculating the position x of each loudspeaker;
setting the order value n of different frequencies, substituting the order value n into a Bessel function, wherein the formula is as follows:
where t is the integral variable, n is the order of the Bessel function, dt is the derivative, representing a small change in t;
calculating the amplitude weight A of each loudspeaker unit and setting the real value J of the Bessel function n (x) Taking an absolute value, namely A= |Jn (x) |; the calculation formula of the position x of each speaker is:n represents the same total number of samples as the number of speaker units, ">Representing the normalized distance value.
Normalized amplitude weight unit: norm_a=a/maxA; converting the amplitude weight to a sound pressure level (dB), a_db=20×log10 (norm_a); calculating a phase offset (Pos) of a speaker unit of the array beam θ, pos=2pi (d/λ) sin (θ×pi/180), where θ is a beam pointing angle and is also a polar angle; calculating the phase weight (P) of each speaker unit, and realizing the direction of the beam pointing angle theta by adding a phase offset: calculating a phase (Φ) using an arctangent function (atan 2), p=atan2 (Im [ Jn (x) ], re [ Jn (x) ]) + Pos x; where Im [ Jn (x) ] represents the imaginary part of the bessel function Jn (x), and Re [ Jn (x) ] represents the real part of the bessel function Jn (x). Where x is an x parameter in Jn (x), and Pos is a phase offset of the speaker unit.
While the foregoing is directed to the preferred embodiments of the present invention, it will be appreciated by those skilled in the art that various modifications and adaptations can be made without departing from the principles of the present invention, and such modifications and adaptations are intended to be comprehended within the scope of the present invention.
Claims (10)
1. An array speaker system, comprising:
the input end analog-to-digital conversion module (1) is used for carrying out analog or digital conversion on an input audio signal source to obtain a digital signal;
the signal matrix module (2) is used for dividing the digital signals to obtain multiple paths of signals;
and the output end processing module is used for carrying out time domain and frequency domain joint processing on the multipath signals to realize the control of the wave field distribution of the loudspeaker array.
2. The array loudspeaker system of claim 1, wherein the output processing module comprises:
a time domain processing module (3) for setting delay time of the multipath signals;
the frequency domain processing module is used for processing each path of signal after delay setting through the high-low pass filter module (4);
the FIR filter module (5) is used for controlling the channel phase and the amplitude of the signal processed by the high-low pass filter module (4);
a multi-channel power amplifier module (6) for amplifying the signals after channel phase and amplitude control;
and an array speaker (7) for converting the amplified and outputted electrical signal into an acoustic wave.
3. A method of implementing an array loudspeaker system according to any one of claims 1 to 2, the method comprising:
determining the number, spacing and frequency parameters of the loudspeakers;
calculating a position vector from each speaker to a reference point based on the array structure;
calculating a unit direction vector of the wave beam according to the requirement;
according to the position vector and the beam direction, calculating the time delay parameter of each loudspeaker by applying a beam forming formula, and according to the operation frequency, calculating the wavelength of the sound wave;
calculating a normalized distance parameter required by the Bessel function based on the position and the wavelength of the loudspeaker, substituting the position of the loudspeaker into the Bessel function, and calculating the amplitude weight and the phase weight of each loudspeaker;
the time domain processing module controls the delay of each loudspeaker according to the time delay parameter, the frequency domain processing module controls the amplitude and the phase of each loudspeaker according to the weight, amplifies the processed driving signals, and the loudspeaker unit converts the amplified electric signals into sound waves.
4. A method of implementing an array speaker system according to claim 3, wherein determining the number of speakers, spacing, frequency parameters comprises:
an array speaker system is set, which is arranged in an array and uniformly spaced, and consists of N identical speaker units, in a specific direction theta 1 and azimuth angleForming a beam thereon;
the radiation frequency of the loudspeaker is f, the sound velocity is c, the cell spacing is d, and a spherical wave model is used,representing the azimuth angle of the array loudspeaker system, wherein in a spherical coordinate system, the sound field is represented by polar coordinates, and theta 2 represents the included angle between the propagation direction of sound waves and the vertical direction; />Representing the angle between the direction of propagation of the sound wave and the reference direction.
5. The method of claim 4, wherein calculating a position vector for each speaker to a reference point based on the array structure, comprises:
calculating d (i) a position vector from the ith element to a reference point of the array, the reference point being provided with a first speaker unit of the array, the array being provided in a cartesian three-dimensional coordinate system, each position being represented by three coordinate values: x, y and z, the position of the ith speaker is (x (i), y (i), z (i)), the geometric center of the array is selected as a reference point, the coordinates are (x (ref), y (ref), z (ref)), the position vector between the two points is represented by the difference of Cartesian coordinates, and the calculation formula of d (i) is:
6. the method of claim 5, wherein calculating the unit direction vector of the beam according to the requirement comprises:
calculating a unit direction vector s, s representing a direction from the reference point to the arrival of the sound source, the direction being defined by the azimuth angle in the spherical coordinate systemAnd a polar angle θ3, for a given azimuth and polar angle, the unit direction vector calculation formula is:
wherein,is azimuth, defined as the angle from the x-axis to the direction vector projected onto the xy-plane, the range is (0, 2pi), θ3 is elevation, defined as the angle from the z-axis to the direction vector, and the range is (0, pi).
7. The method of claim 6, wherein calculating the delay parameter for each speaker using a beamforming formula based on the position vector and the beam direction, and calculating the wavelength of the sound wave based on the operating frequency, comprises:
and calculating the time delay of the array beam pointing angle theta 4, wherein a time delay beam forming formula is as follows:
T(i)＝d(i)×s/c，
wherein, the time delay of the ith element of T (i), d (i) is a vector from the ith element to a reference point of the array, s is a beam unit direction vector, c is sound velocity, and the relation between the wavelength lambda of the sound wave and the frequency f and the sound velocity c is: λ=c/f.
8. The method of claim 7, wherein calculating the normalized distance parameter required for the bessel function based on the speaker location and the wavelength comprises:
parameters required for the bessel function are calculated, the distance norm_d is normalized, and the calculation formula is norm_d=d/λ, where d is the cell spacing and λ is the wavelength.
9. The method of claim 8, wherein substituting the speaker position into the bessel function to calculate the amplitude weight and the phase weight of each speaker comprises:
calculating the position x of each loudspeaker;
setting the order value n of different frequencies, substituting the order value n into a Bessel function, wherein the formula is as follows:
wherein t is an integral variable;
calculating the amplitude weight A of each loudspeaker unit and setting the real value J of the Bessel function n (x) Take the absolute value, i.e., a= |jn (x) |.
10. The method of claim 9, wherein the calculation formula of the position x of each speaker is:n represents the total number of samples, +.>Representing the normalized distance value.
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| CN202311152144.4A Active CN117098045B (en) | 2023-09-07 | 2023-09-07 | Array loudspeaker implementation method |
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| US20070019831A1 (en) * | 2003-06-02 | 2007-01-25 | Yamaha Corporation | Array speaker system |
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| US20190090060A1 (en) * | 2016-03-14 | 2019-03-21 | University Of Southampton | Sound reproduction system |
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| CN113766396A (en) * | 2020-06-05 | 2021-12-07 | 音频风景有限公司 | Loudspeaker control |
| CN116634328A (en) * | 2023-06-07 | 2023-08-22 | 王丁宁 | Broadband ring type phased array sound box system |
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| US20070019831A1 (en) * | 2003-06-02 | 2007-01-25 | Yamaha Corporation | Array speaker system |
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| JP2005210508A (en) * | 2004-01-23 | 2005-08-04 | Yamaha Corp | Cabinet for array speaker and array speaker system |
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| US20190090060A1 (en) * | 2016-03-14 | 2019-03-21 | University Of Southampton | Sound reproduction system |
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