Sound field optimization method and device for multichannel loudspeaker array, equipment and medium

Abstract

The invention provides a sound field optimization method and device of a multichannel loudspeaker array, equipment and a medium, and relates to the technical field of acoustics, to-be-played audio signals are respectively input to each phase compensation filter for phase compensation filtering, and a plurality of phase-compensated audio signals are obtained; the phase compensation filter is determined according to the phase compensation value of the loudspeaker; respectively inputting each phase-compensated audio signal into a corresponding loudspeaker so as to control the multi-channel loudspeaker array to produce sound; acquiring sound field response data in the target area; and when the difference between the sound field response data and the target sound field data does not meet the set sound effect requirement, dynamically adjusting the phase compensation value of the loudspeaker, and repeating the step of phase compensation filtering until the difference between the obtained sound field response data and the target sound field data meets the set sound effect requirement. According to the invention, through a dynamic adjustment mechanism and a closed-loop optimization process, precise regulation and control of the sound field of the loudspeaker array are realized, and the sound effect quality in the target area is significantly improved.

Description

Technical Field

[0001] The present application relates to the field of acoustic technology, and in particular to a sound field optimization method, device, equipment, and medium for a multi-channel speaker array. Background Art

[0002] In recent years, multi-channel speaker array technology, a core component of modern acoustic engineering, has been widely used in professional acoustic scenarios such as high-end cinema systems, intelligent conference spaces, and immersive exhibition halls, aiming to create a high-quality auditory experience for users. However, multi-channel speaker array technology still faces severe challenges in practical application, especially in optimizing the sound field in complex acoustic environments.

[0003] Related technologies typically construct an acoustic model based on the position coordinates of each speaker in a multi-channel speaker array, the preset number of channels, and empirical parameters. The sound field of the multi-channel speaker array is then optimized using a fixed compensation strategy based on the acoustic model. However, even after optimization, poor sound quality still exists. Summary of the Invention

[0004] The present application provides a sound field optimization method, device, equipment and medium for a multi-channel speaker array, which are used to improve the problem of poor sound quality after optimizing the sound field of a multi-channel speaker array in the related art.

[0005] In a first aspect, the present application provides a sound field optimization method for a multi-channel speaker array, comprising:

[0006] Inputting the audio signals to be played into respective phase compensation filters for phase compensation filtering to obtain a plurality of phase-compensated audio signals, wherein the phase compensation filters are determined based on phase compensation values ​​corresponding to speakers in a multi-channel speaker array, wherein the speakers in the multi-channel speaker array are deployed in a target area;

[0007] Input each phase-compensated audio signal to the corresponding speaker to control the sound of the multi-channel speaker array;

[0008] Acquire sound field response data within the target area;

[0009] When the difference between the sound field response data and the target sound field data does not meet the set sound effect requirements, the phase compensation values ​​of the speakers in the multi-channel speaker array are dynamically adjusted, and the audio signals to be played are input into each phase compensation filter for phase compensation filtering to obtain multiple phase-compensated audio signals until the difference between the obtained sound field response data and the target sound field data meets the set sound effect requirements.

[0010] In one possible implementation, dynamically adjusting the phase compensation value of a speaker in a multi-channel speaker array includes adjusting the phase compensation value of the speaker in the multi-channel speaker array based on an adaptive algorithm, the adaptive algorithm including a least mean squares (LMS) algorithm and a recursive least squares (RLS) algorithm.

[0011] In one possible embodiment, the sound field optimization method of a multi-channel speaker array also includes: when switching the adaptive filtering algorithm, a smooth transition mechanism is adopted to gradually adjust the filter weights in the adaptive filtering algorithm through linear interpolation; the smooth transition mechanism includes: before switching the adaptive filtering algorithm, recording the filter weight vector of the current adaptive filtering algorithm; initializing the weight vector of the target adaptive filtering algorithm, and setting a transition time interval; within the transition time interval, through linear interpolation, the filter weights in the adaptive filtering algorithm are smoothly transitioned from the filter weights of the current adaptive filtering algorithm to the weights of the target adaptive filtering algorithm: after the transition is completed, the weight update strategy of the target adaptive filtering algorithm is adopted.

[0012] In one possible embodiment, the initial phase compensation value corresponding to the speaker in the multi-channel speaker array is obtained in the following manner: based on the application scenario characteristics of the target area and the preset acoustic effect requirements, the target sound field data is determined through a pre-established acoustic simulation model, and the target sound field data includes sound pressure level distribution, phase consistency and sound localization accuracy indicators; the Fast Fourier Transform (FFT) algorithm is used to perform spectral analysis on the audio signal to be played, decompose it into different frequency components, and obtain the phase information corresponding to each frequency component; based on the phase information corresponding to each frequency component and the target sound field data, the phase compensation value required for each frequency component on the speaker in the multi-channel speaker array is determined, and the initial phase compensation value corresponding to the speaker in the multi-channel speaker array is obtained.

[0013] In one possible implementation, based on the phase information corresponding to each frequency component and the target sound field data, the phase compensation value required for each frequency component on the speaker in the multi-channel speaker array is determined, including: based on the phase information corresponding to each frequency component and the target sound field data, determining the preliminary phase compensation value required for each frequency component on the speaker in the multi-channel speaker array; based on wave theory and acoustic propagation principles, and taking into account the propagation delay and interference effect of sound in space, introducing a correction coefficient to correct the preliminary phase compensation value required for each frequency component on the speaker in the multi-channel speaker array, so that the sounds emitted by the speakers in the multi-channel speaker array are superimposed in phase or according to the expected phase relationship within the target area, thereby obtaining the phase compensation value required for each frequency component on the speaker in the multi-channel speaker array.

[0014] In one possible embodiment, an acoustic simulation model is established in the following manner: obtaining the number of channels of a multi-channel speaker array, where the channels include expandable channels; obtaining the spatial coordinates corresponding to the speakers in the multi-channel speaker array, where the spatial coordinates are measured by a three-dimensional positioning system; obtaining the frequency response range corresponding to the speakers in the multi-channel speaker array, where the frequency response range is determined based on the average of multiple measurement data within the full frequency band; obtaining phase response data of the speakers in the multi-channel speaker array at different frequencies; and establishing an acoustic simulation model based on the number of channels and the spatial coordinates, frequency response range, and phase response data corresponding to the speakers in the multi-channel speaker array.

[0015] In a second aspect, the present application provides a sound field optimization system, comprising:

[0016] A multi-channel speaker array, comprising a plurality of speakers, for outputting audio signals to be played;

[0017] A sound collection component, including a microphone array, is used to collect sound field response data in the target area and send the sound field response data to a processing device;

[0018] A processing device, configured to execute the sound field optimization method for a multi-channel speaker array according to any one of the first aspects.

[0019] In a third aspect, the present application provides a sound field optimization device for a multi-channel speaker array, comprising:

[0020] a first processing module, configured to input the audio signals to be played into respective phase compensation filters for phase compensation filtering to obtain a plurality of phase-compensated audio signals, wherein the phase compensation filters are determined based on phase compensation values ​​corresponding to speakers in a multi-channel speaker array, wherein the speakers in the multi-channel speaker array are deployed in a target area;

[0021] A second processing module is used to input each phase-compensated audio signal to the corresponding speaker to control the multi-channel speaker array to produce sound;

[0022] An acquisition module, used to acquire sound field response data within a target area;

[0023] The dynamic compensation module is used to dynamically adjust the phase compensation value of the speakers in the multi-channel speaker array when the difference between the sound field response data and the target sound field data does not meet the set sound effect requirements, and execute the steps of inputting the audio signal to be played into each phase compensation filter for phase compensation filtering to obtain multiple phase-compensated audio signals until the difference between the obtained sound field response data and the target sound field data meets the set sound effect requirements.

[0024] In a fourth aspect, the present application provides an electronic device, comprising: a processor, and a memory communicatively connected to the processor;

[0025] Memory for storing computer-executable instructions;

[0026] A processor is configured to execute computer-executable instructions stored in the memory to implement any one of the methods described in the first aspect.

[0027] In a fifth aspect, the present application provides a computer-readable storage medium, which stores computer-executable instructions. When the computer-executable instructions are executed, they are used to implement any one of the methods in the first aspect.

[0028] In a sixth aspect, the present application provides a computer program product, comprising a computer program, which implements the method described in any one of the first aspects when executed.

[0029] The present application provides a sound field optimization method, apparatus, device and medium for a multi-channel speaker array, which inputs the audio signal to be played into each phase compensation filter for phase compensation filtering to obtain multiple phase-compensated audio signals, wherein the phase compensation filter is determined based on the phase compensation value corresponding to the speaker in the multi-channel speaker array, and the speakers in the multi-channel speaker array are deployed in a target area; each phase-compensated audio signal is input into the corresponding speaker to control the sound of the multi-channel speaker array; and sound field response data in the target area is obtained; further, when the difference between the sound field response data and the target sound field data does not meet the set sound effect requirements, the phase compensation value of the speaker in the multi-channel speaker array is dynamically adjusted, and the step of inputting the audio signal to be played into each phase compensation filter for phase compensation filtering to obtain multiple phase-compensated audio signals is executed, until the difference between the obtained sound field response data and the target sound field data meets the set sound effect requirements. During this process, the audio signal to be played is input into the phase compensation filter determined based on the phase compensation value of each speaker for phase compensation filtering, and then the speaker is driven to sound, and the sound field response data in the target area is obtained. When the difference between the sound field response data and the target sound field data does not meet the set sound effect requirements, the speaker phase compensation value is optimized and corrected through the dynamic adjustment mechanism, and the above process is executed repeatedly until the difference between the sound field response data and the target sound field data meets the set sound effect standard. This closed-loop optimization process realizes dynamic and precise control of the sound field of the speaker array, which not only significantly improves the sound quality in the target area, but also ensures that it can highly fit the preset target acoustic effect, greatly enhancing the sound positioning accuracy and clarity; in addition, this dynamic adjustment mechanism can flexibly respond to different scenes and environmental changes, greatly improving the adaptability to complex acoustic scenes, thereby providing a stable, reliable and high-quality listening experience for different application scenarios. BRIEF DESCRIPTION OF THE DRAWINGS

[0030] The accompanying drawings, which are incorporated in and constitute a part of this specification, illustrate embodiments consistent with the present application and, together with the description, serve to explain the principles of the present application.

[0031] Figure 1 A schematic flow chart of a method for optimizing a sound field of a multi-channel speaker array provided by an exemplary embodiment of the present application;

[0032] Figure 2 Another schematic flow chart of a method for optimizing a sound field of a multi-channel speaker array provided by an exemplary embodiment of the present application;

[0033] Figure 3 A schematic structural diagram of a sound field optimization system provided by an exemplary embodiment of the present application;

[0034] Figure 4A schematic structural diagram of a sound field optimization device for a multi-channel speaker array provided by an exemplary embodiment of the present application;

[0035] Figure 5 A schematic structural diagram of an electronic device provided as an exemplary embodiment of the present application.

[0036] The above drawings illustrate specific embodiments of the present application, which will be described in more detail below. These drawings and the textual description are not intended to limit the scope of the present application in any way, but rather to illustrate the concepts of the present application to those skilled in the art by reference to specific embodiments. DETAILED DESCRIPTION

[0037] Exemplary embodiments will be described in detail herein, with examples illustrated in the accompanying drawings. In the following description, when referring to the drawings, identical numerals in different figures represent identical or similar elements, unless otherwise indicated. The embodiments described in the following exemplary embodiments are not intended to represent all embodiments consistent with the present application. Rather, they are merely examples of apparatus and methods consistent with certain aspects of the present application, as detailed in the appended claims.

[0038] The terms "first", "second" etc. in the specification and claims of the present application are used to distinguish similar objects and are not necessarily used to describe a specific order or sequential order. It should be understood that the data used in this way can be interchangeable in appropriate circumstances, so that the embodiments of the present application described herein can, for example, be implemented in an order other than those illustrated or described herein. In addition, the terms "including" and "having" and any of their variations are intended to cover non-exclusive inclusions, for example, the process, system, product or equipment comprising a series of steps or units is not necessarily limited to those steps or units clearly listed, but may include other steps or units that are not clearly listed or that are inherent to these processes, products or equipment.

[0039] It should be noted that the user information (including but not limited to user device information, user personal information, etc.) and data (including but not limited to data used for analysis, stored data, displayed data, etc.) involved in this application are all information and data authorized by the user or fully authorized by all parties, and the collection, use and processing of relevant data must comply with relevant laws, regulations and standards, and provide corresponding operation entrances for users to choose to authorize or refuse.

[0040] In the related art, when a fixed compensation strategy based on an acoustic model is used to optimize the sound field of a multi-channel speaker array, the complex propagation characteristics of sound in space, such as delay, interference effect, and mutual interference between speakers, are ignored. Only simple phase adjustment is performed, and there is a lack of real-time monitoring and dynamic optimization mechanisms. The phase compensation value cannot be adjusted in time according to environmental changes and fluctuations in speaker performance, resulting in the problem of poor sound quality after the sound field of the multi-channel speaker array is optimized.

[0041] In order to solve the above problems, an embodiment of the present application provides a solution for sound field optimization of a multi-channel speaker array. By constructing a phase compensation filter based on the specific phase compensation value of each speaker, the audio signal to be played is input into these filters for phase compensation filtering, so that each audio signal meets the target sound field phase requirements, and multiple phase-compensated audio signals are obtained; then the speaker is driven to sound, and the sound field response data of the target area is collected to evaluate the actual sound field effect; if the difference between the actual sound field response data and the preset target sound field data does not meet the set sound effect standard, the speaker phase compensation value is dynamically adjusted, and the signal filtering process is executed again. Through this closed-loop feedback and iterative optimization control mechanism, the deviation between the actual sound field and the target sound field is gradually converged and eventually eliminated, thereby ensuring high-quality sound output, and thus meeting the diverse acoustic needs in different application scenarios including theaters, conference rooms and exhibition halls.

[0042] The following specific embodiments describe in detail the technical solution of the present application and how the technical solution of the present application solves the above-mentioned technical problems. The following specific embodiments can be combined with each other, and the same or similar concepts or processes may not be repeated in some embodiments. The embodiments of the present application will be described below in conjunction with the accompanying drawings.

[0043] Figure 1 A flow chart of a method for optimizing the sound field of a multi-channel speaker array provided by an exemplary embodiment of the present application. Figure 1 As shown, the sound field optimization method of the multi-channel speaker array includes the following steps:

[0044] S101: Input the audio signal to be played into each phase compensation filter for phase compensation filtering to obtain multiple phase-compensated audio signals, where the phase compensation filter is determined based on the phase compensation value corresponding to the speaker in the multi-channel speaker array, and the speakers in the multi-channel speaker array are deployed in the target area.

[0045] Among them, a multi-channel speaker array refers to an audio system composed of multiple speakers (usually arranged in a specific space), which achieves precise sound field control, sound positioning and immersive audio reproduction through collaborative work; the target areas refer to cinemas, conference rooms and exhibition halls, etc.

[0046] For example, a digital signal processor is used to build a phase compensation filter based on the phase compensation values ​​corresponding to the speakers in the multi-channel speaker array; correspondingly, the audio signals to be played are input into each phase compensation filter for phase compensation filtering to obtain multiple phase-compensated audio signals.

[0047] S102: Input each phase-compensated audio signal to a corresponding speaker to control the multi-channel speaker array to produce sound.

[0048] For example, the processed phase-compensated audio signals are output through a digital-to-analog conversion (D / A) converter and driven by a power amplifier to drive speakers arranged in the target area; each speaker receives a dedicated channel signal, and the synchronization error is controlled within, for example, 100 μs.

[0049] S103: Acquire sound field response data within the target area.

[0050] For example, a sound collection component collects sound field response data within a target area. The sound collection component includes, but is not limited to, a sound collection sensor or a microphone array. The sound collection component then transmits the collected sound field response data within the target area to the sound field optimization system. In turn, the sound field optimization system acquires the sound field response data within the target area.

[0051] S104. When the difference between the sound field response data and the target sound field data does not meet the set sound effect requirements, dynamically adjust the phase compensation values ​​of the speakers in the multi-channel speaker array, and execute the steps of inputting the audio signals to be played into each phase compensation filter for phase compensation filtering to obtain multiple phase-compensated audio signals until the difference between the obtained sound field response data and the target sound field data meets the set sound effect requirements.

[0052] Correspondingly, the sound pressure level deviation between the sound field response data and the target sound field data is calculated; when the sound pressure level deviation is greater than the difference threshold, for example 3dB, the dynamic optimization mechanism is automatically triggered, specifically: the phase compensation value of the speaker in the multi-channel speaker array is dynamically adjusted based on the preset rules, and based on the adjusted phase compensation value, S101 is executed until the difference between the acquired sound field response data and the target sound field data meets the set sound effect requirements.

[0053] The sound field optimization method for a multi-channel speaker array provided in an embodiment of the present application performs phase compensation filtering by inputting the audio signal to be played into a phase compensation filter determined based on the phase compensation value of each speaker, then drives the speaker to make a sound, and obtains the sound field response data in the target area. When the difference between the sound field response data and the target sound field data does not meet the set sound effect requirements, the speaker phase compensation value is optimized and corrected through a dynamic adjustment mechanism, and the above process is executed repeatedly until the difference between the sound field response data and the target sound field data meets the set sound effect standard. This closed-loop optimization process realizes dynamic and precise control of the sound field of the speaker array, which not only significantly improves the sound effect quality in the target area, but also ensures that it can highly fit the preset target acoustic effect, greatly enhancing the sound positioning accuracy and clarity; in addition, this dynamic adjustment mechanism can flexibly respond to different scenes and environmental changes, greatly improving the adaptability to complex acoustic scenes, thereby providing a stable, reliable and high-quality listening experience for different application scenarios.

[0054] In some embodiments, dynamically adjusting the phase compensation value of the speakers in the multi-channel speaker array includes: adjusting the phase compensation value of the speakers in the multi-channel speaker array based on an adaptive algorithm, where the adaptive algorithm includes an LMS algorithm and an RLS algorithm.

[0055] For example, when using a digital signal processing algorithm to perform real-time phase adjustment on an audio signal to be played, a high-performance digital signal processor (DSP) can be selected as the hardware platform. The DSP has powerful digital signal processing capabilities and can quickly execute complex algorithmic operations to meet the needs of real-time phase adjustment. An adaptive filtering algorithm is used to dynamically adjust the filter parameters according to real-time monitored sound field response data, thereby achieving precise adjustment of the phase of the audio signal to be played. In the filter design process, a multi-order infinite impulse response (IIR) filter or a finite impulse response (FIR) filter can be used, combined with optimized filter design methods such as a window function method or a frequency sampling method, so that the filter has good phase characteristics and frequency selectivity. The hardware circuit is optimized to reduce delays and distortions in the signal transmission process, ensuring that the phase-adjusted audio signal can be accurately and quickly sent to each speaker. At the same time, the digital signal processing algorithm is optimized to improve the execution efficiency of the algorithm and reduce the power consumption of the system.

[0056] Specifically, taking the LMS algorithm as an example, its formula is: w(n+1)=w(n)+2μe(n)x(n), where w(n) is the filter coefficient vector at the nth iteration, w(n+1) is the filter coefficient vector at the n+1th iteration, μ is the step size factor used to control the convergence speed and stability, e(n) is the error signal at the nth iteration, and x(n) is the input signal at the nth iteration. The LMS algorithm dynamically adjusts the filter parameters based on the real-time monitored sound field response data. In practical applications, the input signal x(n) can be a characteristic signal extracted from the audio signal to be played, and the error signal e(n) is the difference between the desired target sound field data and the actual monitored sound field response data. By continuously iteratively updating the filter coefficient vector w(n), the error signal e(n) gradually decreases, thereby achieving precise adjustment of the phase of the audio signal to be played. Due to its simple structure and low computational complexity, the LMS algorithm is suitable for scenarios where signal changes are relatively gradual, such as in home theaters. In home theater environments, audio signal changes are generally stable, without excessive fluctuations. The LMS algorithm can quickly converge and stably adjust the phase compensation value, providing users with a good audio experience. However, when encountering scenarios with higher real-time requirements, such as playing movie clips with strong dynamic sound effects, it is possible to switch to the RLS algorithm, which has the characteristics of fast convergence and can better adapt to rapid signal changes.

[0057] For example, the specific implementation of the RLS algorithm is based on the following formula and steps: Assume that w(n) is the filter weight vector at the nth moment, x(n) is the input signal vector at the nth moment, d(n) is the expected signal at the nth moment, y(n) is the filter output signal at the nth moment, e(n) is the error signal at the nth moment, λ is the forgetting factor (0<λ≤1), and P(n) is the inverse matrix of the error covariance matrix. Accordingly, the parameters are first initialized, for example, the filter weight vector w(0)=0, the inverse matrix of the error covariance matrix P(0)=δ -1 I, where δ is a small positive number and I is the identity matrix; then, at each moment n, the following calculation is performed:

[0058] Calculate the filter output: y(n) = w T (n-1)x(n);

[0059] Calculate the error signal: e(n) = d(n) - y(n);

[0060] Calculate the gain vector:

[0061] Update the filter weight vector: w(n) = w(n-1) + k(n)e(n);

[0062] Update the inverse of the error covariance matrix:

[0063] Through the above steps, the RLS algorithm can quickly adjust the filter parameters based on real-time sound field response data, achieve faster phase compensation adjustment, and ensure that users can provide clear and accurate audio effects in dynamic sound scenes.

[0064] Taking into account that when switching the adaptive filtering algorithm, the weight parameters of the filter may suddenly change, thereby causing distortion of the audio signal or the appearance of short-term noise, therefore, based on the above embodiments, in some embodiments, the sound field optimization method of the multi-channel speaker array also includes: when switching the adaptive filtering algorithm, a smooth transition mechanism is adopted to gradually adjust the filter weights in the adaptive filtering algorithm through linear interpolation; the smooth transition mechanism includes: before switching the adaptive filtering algorithm, recording the filter weight vector of the current adaptive filtering algorithm; initializing the weight vector of the target adaptive filtering algorithm, and setting a transition time interval; within the transition time interval, through linear interpolation, the filter weights in the adaptive filtering algorithm are smoothly transitioned from the filter weights of the current adaptive filtering algorithm to the weights of the target adaptive filtering algorithm: after the transition is completed, the weight update strategy of the target adaptive filtering algorithm is adopted.

[0065] For example, when switching from the LMS algorithm to the RLS algorithm, the filter weight vector of the RLS algorithm is not applied immediately, but a smooth transition is adopted. For example, a transition time period T can be set, during which the weight of the RLS algorithm output in the final phase compensation value is gradually increased, while the weight of the LMS algorithm output is gradually reduced. In specific implementation, a linear interpolation method can be used to calculate the weighted sum of the outputs of the RLS algorithm and the LMS algorithm according to the ratio of the current time point to the start and end time points of the transition during the transition period as the final phase compensation value. Similarly, when switching from the RLS algorithm back to the LMS algorithm, a similar smooth transition method is also adopted to gradually adjust the weight parameters of the filter to ensure the continuity and stability of the audio signal and avoid adverse auditory effects caused by algorithm switching.

[0066] In the embodiments of the present application, by dynamically adjusting the speaker phase compensation values ​​in a multi-channel speaker array based on adaptive algorithms (LMS and RLS), the system can quickly respond to environmental changes, optimize the phase compensation values ​​in real time, and reduce sound wave interference and distortion. This not only improves the overall performance of the audio system, but also enhances the user's listening experience, making the sound clearer and more natural, especially in complex acoustic environments. In addition, by using a smooth transition mechanism when switching adaptive filtering algorithms, audio signal distortion and transient noise caused by sudden changes in filter weights caused by algorithm switching are effectively avoided, ensuring that the audio signal maintains continuity and stability during the switching process, making the user almost imperceptible to abnormal changes in sound when switching between different algorithms. At the same time, the smooth transition mechanism provides a buffer for the system to switch between different algorithms, enhancing the system's adaptability to complex acoustic environments and diverse audio content. Whether in home theaters, large performance venues, or other multi-channel audio application scenarios, the speaker phase compensation values ​​can be more accurately and stably adjusted dynamically, improving the overall sound field optimization effect, thereby better meeting the user's pursuit of high-quality audio.

[0067] In some embodiments, the initial phase compensation value corresponding to the speaker in the multi-channel speaker array is obtained in the following manner: based on the application scenario characteristics of the target area and the preset acoustic effect requirements, the target sound field data is determined through a pre-established acoustic simulation model, and the target sound field data includes sound pressure level distribution, phase consistency and sound positioning accuracy indicators; the FFT algorithm is used to perform spectral analysis on the audio signal to be played, and it is decomposed into different frequency components to obtain the phase information corresponding to each frequency component; based on the phase information corresponding to each frequency component and the target sound field data, the phase compensation value required for each frequency component on the speaker in the multi-channel speaker array is determined to obtain the initial phase compensation value corresponding to the speaker in the multi-channel speaker array.

[0068] For example, acoustic simulation software is used to establish an acoustic simulation model. In the model, the size, shape, material properties (such as the sound absorption coefficient of walls, ceilings, and floors) of the target area, as well as parameters such as the position and directivity of the speakers, are set; the visualization function of the acoustic simulation software is used to simulate and analyze different phase compensation schemes. For example, a variety of initial phase compensation settings are tried to observe the sound field distribution and sound propagation path under different schemes; through simulation, the superposition of sound waves in the room under different phase compensation conditions, as well as the phase relationship when the sound reaches different positions, are intuitively seen; after multiple simulations and comparisons, the target sound field data is determined, such as uniform sound field distribution and accurate sound positioning. Among them, the target sound field data specifically includes: for example, in the viewing area, the sound pressure level distribution is required to be between 75dB and 80dB, and the difference between the maximum sound pressure level and the minimum sound pressure level does not exceed 3dB; in terms of phase consistency, the phase difference of the sound emitted by each speaker in the main viewing area is required to be controlled within ±10°; the sound positioning accuracy index is the ability to accurately distinguish the sound source within a range of 15° on the left and right sides of the screen.

[0069] Accordingly, a typical movie audio clip is selected as the audio signal to be played; an audio acquisition device is used to obtain the time domain data of the audio signal, with the sampling frequency set to 44.1kHz and the number of sampling points Q to 1024 to meet the accuracy requirements of spectrum analysis; the collected time domain signal is preprocessed as necessary, such as removing the DC component and filtering to remove noise, to improve the accuracy of spectrum analysis; further, the FFT algorithm is used to perform spectrum analysis on the preprocessed time domain signal, and the FFT algorithm satisfies the following formula:

[0070]

[0071] Among them, x(q) is the time domain signal, q represents the time domain sampling point number, Q is the number of sampling points, X(k) is the frequency domain signal, k represents the frequency domain sampling point number, is the rotation factor.

[0072] Furthermore, q takes values ​​from 0 to Q-1, representing the calculation of Q time domain sampling points; k takes values ​​from 0 to Q-1, obtaining Q frequency domain sampling point results; Q is the number of sampling points, which determines the resolution and computational complexity of the spectrum analysis; the time domain signal is decomposed into different frequency components through the FFT algorithm to obtain the phase information corresponding to each frequency component. For example, for 1024 time domain sampling points, 1024 frequency domain sampling point results are obtained after FFT calculation, and each frequency domain sampling point corresponds to a specific frequency component and its phase information; based on the phase information corresponding to each frequency component and the previously determined target sound field data, a comprehensive analysis is performed to determine the phase compensation value required for each frequency component on the speaker in the multi-channel speaker array, and obtain the initial phase compensation value corresponding to the speaker in the multi-channel speaker array. For example, for the left front speaker, at a frequency component of 100Hz, a phase compensation of 15° may be required; for the right front speaker, at a frequency component of 1000Hz, a phase compensation of -8° may be required, and so on. These initial phase compensation values ​​will serve as the basis for subsequent dynamic adjustments, providing a starting point for achieving the target acoustic effect.

[0073] In some embodiments, based on the phase information corresponding to each frequency component and the target sound field data, the phase compensation value required for each frequency component on the speaker in the multi-channel speaker array is determined, including: based on the phase information corresponding to each frequency component and the target sound field data, the preliminary phase compensation value required for each frequency component on the speaker in the multi-channel speaker array is determined; based on the wave theory and the principle of acoustic propagation, and taking into account the propagation delay and interference effect of sound in space, a correction coefficient is introduced to correct the preliminary phase compensation value required for each frequency component on the speaker in the multi-channel speaker array, so that the sounds emitted by the speakers in the multi-channel speaker array are superimposed in phase or according to the expected phase relationship within the target area, thereby obtaining the phase compensation value required for each frequency component on the speaker in the multi-channel speaker array.

[0074] For example, based on wave theory and acoustic propagation principles, the propagation delay and interference effects of sound in space are considered. For example, for low-frequency components, due to their longer wavelengths, they are relatively less affected by the environment during propagation, so the impact of the initial phase difference between each speaker on the superposition of the low-frequency sound field needs to be considered. For high-frequency components, due to their shorter wavelengths, they are more susceptible to interference from factors such as reflection or scattering, and the phase compensation value needs to be adjusted more accurately to ensure accurate positioning of high-frequency sounds. Accordingly, by measuring the propagation distance of each path from the speaker to the center of the target area, the propagation delay is calculated in combination with the speed of sound, and factors such as the reflection coefficient of the target area are considered to introduce a correction coefficient. For low-frequency components, the correction coefficient mainly corrects the propagation delay, with a value range of, for example, 0.9-1.1. For high-frequency components, the correction coefficient also needs to take interference into account, with a value range of, for example, 0.8-1.2. The initial phase compensation value is corrected by the correction coefficient to obtain the phase compensation value required for each frequency component on the speaker in the multi-channel speaker array, providing a basis for subsequent dynamic adjustment.

[0075] The embodiment of the present application obtains target sound field data with the help of an acoustic simulation model based on the application scenario of the target area and the preset acoustic effect requirements, and performs spectral analysis on the audio signal to be played in combination with the FFT algorithm to obtain phase information and determine the initial phase compensation value, which can accurately adapt to the acoustic requirements of different scenarios; further, based on the wave theory and the principle of acoustic propagation, a correction coefficient is introduced to take into account the propagation delay and interference effect, so that the sound emitted by each speaker can be superimposed in the same phase or expected phase relationship within the target area, which not only improves the accuracy of sound positioning, allowing the audience to clearly distinguish the direction of the sound source, but also enhances the uniformity of the sound pressure level distribution, avoids the situation where the sound fluctuates, and effectively improves the overall acoustic effect, thereby bringing users a better quality and immersive sound experience.

[0076] Considering that in the related art, in the initial parameter setting, the speaker position coordinates are roughly measured with poor accuracy, and the number of channels is determined only by the number of fixed audio source channels, lacking flexibility and expansion. Therefore, in some embodiments, the acoustic simulation model is established in the following manner: obtaining the number of channels of the multi-channel speaker array, the channels including expandable channels; obtaining the spatial coordinates corresponding to the speakers in the multi-channel speaker array, the spatial coordinates being measured by a three-dimensional positioning system; obtaining the frequency response range corresponding to the speakers in the multi-channel speaker array, the frequency response range being determined based on the average of multiple measurement data within the full frequency band; obtaining the phase response data of the speakers in the multi-channel speaker array at different frequencies; and establishing the acoustic simulation model based on the number of channels and the spatial coordinates corresponding to the speakers in the multi-channel speaker array, the frequency response range, and the phase response data at different frequencies.

[0077] For example, when determining the number of channels in a multi-channel speaker array, not only the current number of audio source channels is considered, but also the reserved expansion space to allow for subsequent channel additions based on actual needs. A high-precision three-dimensional positioning system, such as a three-dimensional laser positioning system, is used to determine the position coordinates of each speaker. This positioning system utilizes the principle of laser ranging. By emitting a laser beam at the speaker surface and measuring the time it takes for the reflected light to travel, it can accurately calculate the speaker's three-dimensional coordinates in space. This effectively reduces acoustic effect deviations caused by position measurement errors and provides an accurate spatial data foundation for subsequent acoustic simulations. Accordingly, professional audio test equipment, such as an audio analyzer, is used to perform multiple frequency response measurements on each speaker over the full frequency range, for example, 20Hz-20kHz. The audio analyzer's signal output is connected to the speaker's input, and the measurement output is connected to the speaker's acoustic measurement point (usually 1 meter directly in front of the speaker). During each measurement, the audio analyzer outputs a sine wave signal of a different frequency and measures the speaker's sound pressure level response at that frequency. Each frequency point is measured, for example, five times, and the average of these five measurements is taken as the frequency response value for that frequency point. For example, at a frequency point of 100Hz, the five measured sound pressure levels are 85dB, 86dB, 84dB, 85dB, and 86dB, respectively, resulting in an average of 85.2dB. This method generates a frequency response curve for each speaker over the full frequency range, thereby determining its frequency response range and improving the accuracy of the frequency response data. Correspondingly, an audio analyzer is also used to obtain the phase response data of the speaker at different frequencies. During the measurement process, the audio analyzer not only measures the sound pressure level response, but also simultaneously measures the phase change of the signal. For each speaker, measurements are performed at a certain frequency interval (such as 10Hz) within the frequency band of 20Hz-20kHz. For example, at a frequency of 1kHz, the phase difference of the speaker output signal relative to the input signal is measured. Since the actual measurement process may be affected by factors such as environmental noise and electromagnetic interference, resulting in noise and outliers in the obtained phase response data, these data need to be preprocessed. For example, a median filter algorithm is used to remove obvious noise and outliers, and then a sliding average filter algorithm is used to smooth the data to further reduce the interference of random noise and make the phase response curve smoother.

[0078] Furthermore, based on the obtained number of channels, such as 7.1 channels and the ability to be expanded to 16 channels, the spatial coordinates corresponding to the speakers in the multi-channel speaker array, the frequency response range (the frequency response curve of each speaker determined based on the average of multiple measurement data of the full frequency band) and the phase response data at different frequencies (the smoothed phase response curve after preprocessing), an acoustic simulation model is established using acoustic simulation software. Specifically, in the acoustic simulation software, the spatial coordinates of the speakers are first input to construct a virtual speaker array layout; the frequency response range and phase response data of each speaker are imported into the software as the acoustic characteristic parameters of the speaker; the relevant parameters of the acoustic simulation environment are set, such as size, wall material and audience position; through the acoustic algorithm within the software, the propagation, reflection, interference and other phenomena of sound in space are simulated, and acoustic indicators such as sound pressure level distribution, phase consistency and sound positioning accuracy at different positions are calculated, thereby obtaining an acoustic simulation model that can accurately simulate the acoustic effect of the multi-channel speaker array in the actual environment, providing strong support for subsequent speaker layout optimization, parameter adjustment and system debugging.

[0079] In the embodiment of the present application, a high-precision loudspeaker spatial coordinate is obtained through a three-dimensional positioning system, and then combined with the frequency response range determined by the average of multiple measurements across the entire frequency band and the pre-processed phase response data, a highly accurate acoustic simulation model is constructed, thereby providing a solid and reliable foundation for phase compensation calculation and sound field optimization; in addition, the expandable channel design reserved in the model enables it to flexibly adapt to different channel requirements, thereby enhancing flexibility and versatility.

[0080] Figure 2 Another flow chart of the sound field optimization method of the multi-channel speaker array provided by the exemplary embodiment of the present application. Figure 2 As shown, the sound field optimization method of the multi-channel speaker array includes the following steps:

[0081] S201. Obtain spatial coordinates, frequency response range, and phase response data at different frequencies corresponding to speakers in a multi-channel speaker array, and establish an acoustic simulation model based on the number of channels and the spatial coordinates, frequency response range, and phase response data at different frequencies corresponding to the speakers in the multi-channel speaker array.

[0082] The spatial coordinates are measured using a three-dimensional positioning system, and the frequency response range is determined based on the average of multiple measurement data within the full frequency band.

[0083] S202. Based on the application scenario characteristics of the target area and the preset acoustic effect requirements, the target sound field data is determined through a pre-established acoustic simulation model. The target sound field data includes sound pressure level distribution, phase consistency, and sound localization accuracy indicators.

[0084] S203 , using a fast FFT algorithm to perform spectrum analysis on the audio signal to be played, decomposing it into different frequency components, and obtaining phase information corresponding to each frequency component.

[0085] S204 : Determine the phase compensation value required for each frequency component on a speaker in the multi-channel speaker array based on the phase information corresponding to each frequency component and the target sound field data, and obtain an initial phase compensation value corresponding to the speaker in the multi-channel speaker array.

[0086] S205: Inputting the audio signal to be played into each phase compensation filter for phase compensation filtering to obtain a plurality of phase-compensated audio signals, wherein the phase compensation filters are determined based on phase compensation values ​​corresponding to speakers in a multi-channel speaker array, where the speakers in the multi-channel speaker array are deployed in a target area;

[0087] S206 , inputting each phase-compensated audio signal to a corresponding speaker to control the multi-channel speaker array to produce sound.

[0088] S207: Acquire sound field response data within the target area.

[0089] For example, a microphone array deployed in the target area collects sound field response data in the target area, and sends the sound field response data to the processing device; correspondingly, the processing device obtains the sound field response data in the target area.

[0090] S208. When the difference between the sound field response data and the target sound field data does not meet the set sound effect requirements, dynamically adjust the phase compensation values ​​of the speakers in the multi-channel speaker array, and execute the steps of inputting the audio signals to be played into each phase compensation filter for phase compensation filtering to obtain multiple phase-compensated audio signals, until the difference between the obtained sound field response data and the target sound field data meets the set sound effect requirements.

[0091] For example, the phase compensation values ​​of the speakers in the multi-channel speaker array are adjusted based on an adaptive algorithm, where the adaptive algorithm includes an LMS algorithm and an RLS algorithm.

[0092] Accordingly, when switching the adaptive filtering algorithm, a smooth transition mechanism is adopted to gradually adjust the filter weights in the adaptive filtering algorithm through linear interpolation; wherein, the smooth transition mechanism includes: before switching the adaptive filtering algorithm, recording the filter weight vector of the current adaptive filtering algorithm; initializing the weight vector of the target adaptive filtering algorithm, and setting a transition time interval; within the transition time interval, through linear interpolation, making the filter weights in the adaptive filtering algorithm smoothly transition from the filter weights of the current adaptive filtering algorithm to the weights of the target adaptive filtering algorithm: after the transition is completed, adopting the weight update strategy of the target adaptive filtering algorithm.

[0093] In summary, this application has at least the following advantages:

[0094] 1. By inputting the audio signal to be played into the phase compensation filter determined based on the phase compensation value of each speaker for phase compensation filtering, and then driving the speaker to sound, and obtaining the sound field response data in the target area, when the difference between the sound field response data and the target sound field data does not meet the set sound effect requirements, the speaker phase compensation value is optimized and corrected through the dynamic adjustment mechanism, and the above process is executed repeatedly until the difference between the sound field response data and the target sound field data meets the set sound effect standard. This closed-loop optimization process realizes dynamic and precise control of the sound field of the speaker array, which not only significantly improves the sound quality in the target area, but also ensures that it can highly fit the preset target acoustic effect, greatly enhancing the sound positioning accuracy and clarity; in addition, this dynamic adjustment mechanism can flexibly respond to different scenes and environmental changes, greatly improving the adaptability to complex acoustic scenes, thereby providing a stable, reliable and high-quality listening experience for different application scenarios.

[0095] Second, by dynamically adjusting speaker phase compensation in a multi-channel speaker array using adaptive algorithms (LMS and RLS), the system rapidly responds to environmental changes, optimizing phase compensation in real time and reducing acoustic interference and distortion. This not only improves the overall performance of the audio system but also enhances the user's listening experience, making the sound clearer and more natural, especially in complex acoustic environments. Furthermore, a smooth transition mechanism is implemented when switching between adaptive filtering algorithms, effectively avoiding audio signal distortion and transient noise caused by sudden changes in filter weights during algorithm switching. This ensures continuity and stability of the audio signal during switching, making the user virtually imperceptible to any changes in the sound when switching between algorithms. Furthermore, the smooth transition mechanism provides a buffer for switching between algorithms, enhancing the system's adaptability to complex acoustic environments and diverse audio content. Whether in home theaters, large performance venues, or other multi-channel audio scenarios, the system enables more precise and stable dynamic adjustment of speaker phase compensation, improving overall sound field optimization and better satisfying users' pursuit of high-quality audio.

[0096] 3. Based on the application scenarios of the target area and the preset acoustic effect requirements, the target sound field data is obtained with the help of an acoustic simulation model, and the frequency spectrum analysis of the audio signal to be played is combined with the FFT algorithm to obtain phase information and determine the initial phase compensation value, which can accurately adapt to the acoustic requirements of different scenarios; further, based on the wave theory and the principle of acoustic propagation, the correction coefficient is introduced to take into account the propagation delay and interference effects, so that the sound emitted by each speaker can be superimposed in the same phase or expected phase relationship in the target area, which not only improves the accuracy of sound positioning, allowing the audience to clearly distinguish the direction of the sound source, but also enhances the uniformity of the sound pressure level distribution, avoids the situation where the sound fluctuates, and effectively improves the overall acoustic effect, thereby bringing users a better quality and immersive sound experience.

[0097] Fourth, a highly accurate acoustic simulation model is constructed by obtaining high-precision loudspeaker spatial coordinates through a three-dimensional positioning system. This is combined with the frequency response range determined by averaging multiple measurements across the entire frequency band and pre-processed phase response data. This provides a solid and reliable foundation for phase compensation calculations and sound field optimization. Furthermore, the model's scalable sound channel design allows for flexible adaptation to varying sound channel requirements, enhancing flexibility and versatility.

[0098] Figure 3 A structural diagram of a sound field optimization system provided by an exemplary embodiment of the present application. Figure 3 As shown, the sound field optimization system 30 includes a multi-channel speaker array 31, a sound collection component 32 and a processing device 33, wherein:

[0099] A multi-channel speaker array 31, comprising a plurality of speakers, for outputting audio signals to be played;

[0100] a sound collection component 32 comprising a microphone array for collecting sound field response data within a target area and sending the sound field response data to a processing device;

[0101] The processing device 33 is configured to execute the sound field optimization method for a multi-channel speaker array according to any one of the above embodiments.

[0102] Figure 4 A schematic diagram of a sound field optimization device for a multi-channel speaker array provided by an exemplary embodiment of the present application. Figure 4 As shown, the sound field optimization device 40 for a multi-channel speaker array includes a first processing module 41, a second processing module 42, an acquisition module 43 and a dynamic compensation module 44, wherein:

[0103] a first processing module 41 for inputting the audio signals to be played into respective phase compensation filters for phase compensation filtering to obtain a plurality of phase-compensated audio signals, wherein the phase compensation filters are determined based on phase compensation values ​​corresponding to speakers in a multi-channel speaker array, wherein the speakers in the multi-channel speaker array are deployed in a target area;

[0104] The second processing module 42 is used to input each phase-compensated audio signal to the corresponding speaker to control the multi-channel speaker array to produce sound;

[0105] An acquisition module 43 is used to acquire sound field response data within a target area;

[0106] The dynamic compensation module 44 is used to dynamically adjust the phase compensation value of the speakers in the multi-channel speaker array when the difference between the sound field response data and the target sound field data does not meet the set sound effect requirements, and execute the steps of inputting the audio signal to be played into each phase compensation filter for phase compensation filtering to obtain multiple phase-compensated audio signals until the difference between the obtained sound field response data and the target sound field data meets the set sound effect requirements.

[0107] In a possible implementation, the dynamic compensation module 44 may be specifically configured to adjust the phase compensation value of the speakers in the multi-channel speaker array based on an adaptive algorithm, where the adaptive algorithm includes an LMS algorithm and an RLS algorithm.

[0108] In one possible implementation, the dynamic compensation module 44 can also be used to: when switching the adaptive filtering algorithm, adopt a smooth transition mechanism to gradually adjust the filter weights in the adaptive filtering algorithm through linear interpolation; the smooth transition mechanism includes: before switching the adaptive filtering algorithm, recording the filter weight vector of the current adaptive filtering algorithm; initializing the weight vector of the target adaptive filtering algorithm, and setting a transition time interval; within the transition time interval, through linear interpolation, the filter weights in the adaptive filtering algorithm are smoothly transitioned from the filter weights of the current adaptive filtering algorithm to the weights of the target adaptive filtering algorithm: after the transition is completed, the weight update strategy of the target adaptive filtering algorithm is adopted.

[0109] In one possible implementation, the first processing module 41 can be specifically used to: determine the target sound field data through a pre-established acoustic simulation model based on the application scenario characteristics of the target area and the preset acoustic effect requirements, the target sound field data including sound pressure level distribution, phase consistency and sound positioning accuracy indicators; use the FFT algorithm to perform spectral analysis on the audio signal to be played, decompose it into different frequency components, and obtain the phase information corresponding to each frequency component; based on the phase information corresponding to each frequency component and the target sound field data, determine the phase compensation value required for each frequency component on the speaker in the multi-channel speaker array, and obtain the initial phase compensation value corresponding to the speaker in the multi-channel speaker array.

[0110] In one possible implementation, the first processing module 41 may also be used to: determine the preliminary phase compensation value required for each frequency component on the speaker in the multi-channel speaker array based on the phase information and target sound field data corresponding to each frequency component; introduce a correction coefficient based on wave theory and acoustic propagation principles, and taking into account the propagation delay and interference effect of sound in space, and correct the preliminary phase compensation value required for each frequency component on the speaker in the multi-channel speaker array, so that the sounds emitted by the speakers in the multi-channel speaker array are superimposed in phase or according to the expected phase relationship within the target area, thereby obtaining the phase compensation value required for each frequency component on the speaker in the multi-channel speaker array.

[0111] In one possible implementation, an acoustic simulation model is established in the following manner: obtaining the number of channels of a multi-channel speaker array, where the channels include expandable channels; obtaining the spatial coordinates corresponding to the speakers in the multi-channel speaker array, where the spatial coordinates are measured by a three-dimensional positioning system; obtaining the frequency response range corresponding to the speakers in the multi-channel speaker array, where the frequency response range is determined based on the average of multiple measurement data within the full frequency band; obtaining phase response data of the speakers in the multi-channel speaker array at different frequencies; and establishing an acoustic simulation model based on the number of channels and the spatial coordinates, frequency response range, and phase response data corresponding to the speakers in the multi-channel speaker array.

[0112] The sound field optimization device for a multi-channel speaker array provided in the embodiment of the present application can implement the technical solution shown in the above-mentioned sound field optimization method embodiment of a multi-channel speaker array. Its implementation principle and beneficial effects are similar and will not be repeated here.

[0113] It should be noted that it should be understood that the division of the various modules of the above device is merely a division of logical functions. In actual implementation, they can be fully or partially integrated into one physical entity, or they can be physically separated. Moreover, these modules can all be implemented in the form of software called by processing elements; they can also all be implemented in the form of hardware; some modules can also be implemented in the form of software called by processing elements, and some modules can be implemented in the form of hardware. For example, the dynamic compensation module can be a separately established processing element, or it can be integrated into a chip of the above device. In addition, it can also be stored in the memory of the above device in the form of program code, and called by a processing element of the above device to perform the functions of the above dynamic compensation module. The implementation of other modules is similar. In addition, these modules can all or partly be integrated together, or they can be implemented independently. The processing element here can be an integrated circuit with signal processing capabilities. In the implementation process, each step of the above method or each of the above modules can be completed by the hardware integrated logic circuit in the processor element or software instructions.

[0114] For example, the above modules can be one or more integrated circuits configured to implement the above methods, such as one or more application-specific integrated circuits (ASICs), one or more microprocessors (DSPs), or one or more field programmable gate arrays (FPGAs). For another example, when a module is implemented by scheduling program code through a processing element, the processing element can be a general-purpose processor, such as a central processing unit (CPU) or other processor that can call program code. For another example, these modules can be integrated together and implemented in the form of a system-on-a-chip (SOC).

[0115] In the above embodiments, it can be implemented in whole or in part by software, hardware, firmware or any combination thereof. When software is used for implementation, it can be implemented in whole or in part in the form of a computer program product. The computer program product includes one or more computer instructions. When the computer instructions are loaded and executed on a computer, the process or function according to the embodiment of the present application is generated in whole or in part. The computer can be a general-purpose computer, a special-purpose computer, a computer network, or other programmable devices. The computer instructions can be stored in a computer-readable storage medium or transmitted from one computer-readable storage medium to another computer-readable storage medium. For example, the computer instructions can be transmitted from a website, computer, server or data center to another website, computer, server or data center by wired (such as coaxial cable, optical fiber, digital subscriber line (DSL)) or wireless (such as infrared, wireless, microwave, etc.) mode. The computer-readable storage medium can be any available medium that a computer can access or a data storage device such as a server or data center that includes one or more available media integrations. Available media may be magnetic media (eg, floppy disks, hard disks, magnetic tapes), optical media (eg, Digital Video Discs (DVDs)), or semiconductor media (eg, solid state disks (SSDs)).

[0116] Figure 5 This is a schematic diagram of the structure of an electronic device provided by an exemplary embodiment of the present application. Figure 5 As shown, the electronic device 50 of this embodiment includes:

[0117] At least one processor 51; and a memory 52 communicatively connected to the at least one processor;

[0118] The memory 52 stores instructions that can be executed by the at least one processor 51 , and the instructions are executed by the at least one processor 51 to enable the electronic device to execute the method as described in any of the above embodiments.

[0119] Optionally, the memory 52 may be independent or integrated with the processor 51 .

[0120] The memory 52 may include a high-speed random access memory (RAM), and may also include a non-volatile memory (non-volatile memory), such as at least one disk memory.

[0121] The processor 51 may be a central processing unit (CPU), an application-specific integrated circuit (ASIC), or one or more integrated circuits configured to implement the embodiments of the present application. Specifically, when implementing the sound field optimization method for a multi-channel speaker array described in the aforementioned method embodiment, the electronic device may be, for example, an electronic device with processing capabilities, such as a server.

[0122] Optionally, the electronic device may further include a communication interface 53. In a specific implementation, if the communication interface 53, the memory 52, and the processor 51 are implemented independently, the communication interface 53, the memory 52, and the processor 51 may be interconnected via a bus and communicate with each other. The bus may be an Industry Standard Architecture (ISA) bus, a Peripheral Component Interconnect (PCI) bus, or an Extended Industry Standard Architecture (EISA) bus. Buses can be divided into address buses, data buses, control buses, etc., but this does not mean that there is only one bus or only one type of bus.

[0123] Optionally, in a specific implementation, if the communication interface 53, the memory 52 and the processor 51 are integrated on a chip, the communication interface 53, the memory 52 and the processor 51 can complete communication through an internal interface.

[0124] The implementation principle and technical effects of the electronic device provided in this embodiment can be found in the aforementioned embodiments and will not be described in detail here.

[0125] An embodiment of the present application also provides a computer-readable storage medium, which stores computer execution instructions. When the computer execution instructions are executed, they are used to implement the method steps in the above method embodiment. The specific implementation method and technical effects are similar and will not be repeated here.

[0126] The computer-readable storage medium may be implemented by any type of volatile or non-volatile storage device or a combination thereof, such as static random access memory (SRAM), electrically erasable programmable read-only memory (EEPROM), erasable programmable read-only memory (EPROM), programmable read-only memory (PROM), read-only memory (ROM), magnetic storage, flash memory, magnetic disk, or optical disk. The computer-readable storage medium may be any available medium that can be accessed by a general-purpose or special-purpose computer.

[0127] An exemplary readable storage medium is coupled to a processor, enabling the processor to read information from and write information to the readable storage medium. Alternatively, the readable storage medium may be a component of the processor. The processor and the readable storage medium may be located in an application-specific integrated circuit. Alternatively, the processor and the readable storage medium may be discrete components within a sound field optimization device for a multi-channel speaker array.

[0128] An embodiment of the present application also provides a computer program product, including a computer program. When the computer program is executed, the method steps in the above method embodiment are implemented. The specific implementation method and technical effects are similar and will not be repeated here.

[0129] Those skilled in the art will readily appreciate other embodiments of the present application after considering the specification and practicing the invention disclosed herein. This application is intended to cover any variations, uses, or adaptations of the present application that follow the general principles of the present application and include common knowledge or customary techniques in the art not disclosed herein. The description and examples are to be considered as exemplary only, and the true scope and spirit of the present application are indicated by the following claims.

[0130] It should be understood that the present application is not limited to the exact structure described above and shown in the drawings, and that various modifications and changes may be made without departing from the scope thereof. The scope of the present application is limited only by the appended claims.

Claims

1. A sound field optimization method for a multi-channel speaker array, characterized in that: include: Inputting the audio signals to be played into respective phase compensation filters for phase compensation filtering to obtain a plurality of phase-compensated audio signals, wherein the phase compensation filters are determined based on phase compensation values ​​corresponding to speakers in a multi-channel speaker array, wherein the speakers in the multi-channel speaker array are deployed in a target area; Inputting each of the phase-compensated audio signals into a corresponding speaker to control the multi-channel speaker array to produce sound; Acquiring sound field response data within the target area; When the difference between the sound field response data and the target sound field data does not meet the set sound effect requirements, the phase compensation values ​​of the speakers in the multi-channel speaker array are dynamically adjusted, and the steps of inputting the audio signals to be played into each phase compensation filter for phase compensation filtering to obtain multiple phase-compensated audio signals are performed until the difference between the acquired sound field response data and the target sound field data meets the set sound effect requirements.

2. The sound field optimization method according to claim 1, characterized in that: The dynamically adjusting the phase compensation value of the speaker in the multi-channel speaker array includes: The phase compensation values ​​of the speakers in the multi-channel speaker array are adjusted based on an adaptive algorithm, wherein the adaptive algorithm includes a least mean square (LMS) algorithm and a recursive least square (RLS) algorithm.

3. The sound field optimization method according to claim 2, characterized in that: Also includes: When switching the adaptive filtering algorithm, a smooth transition mechanism is adopted to gradually adjust the filter weights in the adaptive filtering algorithm through linear interpolation; The smooth transition mechanism includes: Before switching the adaptive filtering algorithm, recording the filter weight vector of the current adaptive filtering algorithm; Initialize the weight vector of the target adaptive filtering algorithm and set the transition time interval; In the transition time interval, the filter weights in the adaptive filtering algorithm are smoothly transitioned from the filter weights of the current adaptive filtering algorithm to the weights of the target adaptive filtering algorithm through linear interpolation: After the transition is completed, the weight update strategy of the target adaptive filtering algorithm is adopted.

4. The sound field optimization method according to any one of claims 1 to 3, characterized in that: The initial phase compensation values ​​corresponding to the speakers in the multi-channel speaker array are obtained in the following manner: Based on the application scenario characteristics of the target area and the preset acoustic effect requirements, the target sound field data is determined by a pre-established acoustic simulation model, and the target sound field data includes sound pressure level distribution, phase consistency, and sound localization accuracy indicators; The Fast Fourier Transform (FFT) algorithm is used to perform spectrum analysis on the audio signal to be played, decompose it into different frequency components, and obtain the phase information corresponding to each frequency component; Based on the phase information corresponding to each frequency component and the target sound field data, the phase compensation value required for each frequency component on the speaker in the multi-channel speaker array is determined to obtain the initial phase compensation value corresponding to the speaker in the multi-channel speaker array.

5. The sound field optimization method according to claim 4, characterized in that: The determining, based on the phase information corresponding to each frequency component and the target sound field data, a phase compensation value required for each frequency component on a speaker in the multi-channel speaker array includes: determining, based on the phase information corresponding to each frequency component and the target sound field data, a preliminary phase compensation value required for each frequency component on a speaker in the multi-channel speaker array; Based on wave theory and acoustic propagation principles, and taking into account the propagation delay and interference effects of sound in space, a correction coefficient is introduced to correct the preliminary phase compensation value required for each frequency component on the speakers in the multi-channel speaker array. This allows the sounds emitted by the speakers in the multi-channel speaker array to be superimposed in phase or according to a desired phase relationship within the target area, thereby obtaining the phase compensation value required for each frequency component on the speakers in the multi-channel speaker array.

6. The sound field optimization method according to claim 4, characterized in that: The acoustic simulation model is established in the following way: Obtaining the number of channels of the multi-channel speaker array, where the channels include expandable channels; Obtaining spatial coordinates corresponding to speakers in the multi-channel speaker array, where the spatial coordinates are measured by a three-dimensional positioning system; Obtaining a frequency response range corresponding to a speaker in the multi-channel speaker array, where the frequency response range is determined based on an average of multiple measurement data within a full frequency band; Obtaining phase response data of speakers in the multi-channel speaker array at different frequencies; The acoustic simulation model is established based on the number of channels and the spatial coordinates, frequency response range, and phase response data at different frequencies corresponding to the speakers in the multi-channel speaker array.

7. A sound field optimization system, characterized in that: include: A multi-channel speaker array, comprising a plurality of speakers, for outputting audio signals to be played; A sound collection component, comprising a microphone array, for collecting sound field response data within a target area and sending the sound field response data to a processing device; The processing device is used to execute the sound field optimization method of the multi-channel speaker array according to any one of claims 1 to 6.

8. A sound field optimization device for a multi-channel speaker array, characterized in that: include: a first processing module, configured to input the audio signals to be played into respective phase compensation filters for phase compensation filtering to obtain a plurality of phase-compensated audio signals, wherein the phase compensation filters are determined based on phase compensation values ​​corresponding to speakers in a multi-channel speaker array, wherein the speakers in the multi-channel speaker array are deployed in a target area; a second processing module, configured to input each of the phase-compensated audio signals into a corresponding speaker, so as to control the multi-channel speaker array to produce sound; An acquisition module, configured to acquire sound field response data within the target area; A dynamic compensation module is used to dynamically adjust the phase compensation values ​​of the speakers in the multi-channel speaker array when the difference between the sound field response data and the target sound field data does not meet the set sound effect requirements, and execute the step of inputting the audio signal to be played into each phase compensation filter for phase compensation filtering to obtain multiple phase-compensated audio signals until the difference between the acquired sound field response data and the target sound field data meets the set sound effect requirements.

9. An electronic device, characterized in that: include: a processor, and a memory communicatively connected to the processor; The memory is used to store computer-executable instructions; The processor is configured to execute the computer-executable instructions to implement the method according to any one of claims 1 to 6.

10. A computer-readable storage medium, characterized in that The computer-readable storage medium stores computer-executable instructions, which are used to implement the method according to any one of claims 1 to 6 when executed.

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