Playing frequency response adaptive equalization method and device

Abstract

The invention relates to the technical field of playing frequency response equalization, and discloses a playing frequency response adaptive equalization method and device, and the method comprises the steps: obtaining an original playing signal of a loudspeaker and an audio signal collected by a microphone; designing a room transfer function between the original playing signal and the audio signal by using a preset adaptive filtering algorithm; estimating a frequency response compensation curve according to the room transfer function; and based on the frequency response compensation curve, designing a corresponding IIR filter to carry out frequency response compensation on the original playing signal. The method can achieve the effects of improving the measurement precision of the frequency response, being simple in measurement method, enhancing the robustness, and remarkably improving the frequency response performance of the loudspeaker system.

Description

Technical Field

[0001] The present application relates to the technical field of playback frequency response equalization, and in particular to a playback frequency response adaptive equalization method and device. Background Art

[0002] In the speaker system, playback frequency response equalization technology is mainly used to improve the frequency response of the audio system, making the sound more uniform and flat at each frequency.

[0003] At present, one solution is to obtain the frequency response of the system through spectrum analysis of the pulse signal played by the speaker, decompose the frequency response to obtain the minimum phase part and then invert it to compensate for the amplitude response of the speaker system. However, its implementation is complex, especially when dealing with high-order systems. The frequency response decomposition process involves complex mathematical operations and high computational complexity. It is also difficult to measure. Accurately measuring the pulse response requires professional equipment and measurement methods, and has strict requirements on the signal-to-noise ratio of the measurement environment. Another solution is to directly calculate the spectral energy of the signal based on the white noise or swept frequency signal played by the speaker, obtain the amplitude-frequency response of the speaker system, and then compensate the amplitude response of the system through a frequency domain equalizer. Although its measurement and calculation methods are relatively simple, it is easily affected by interference noise in the environment, resulting in measurement deviations, and thus poor robustness.

[0004] With respect to the above-mentioned related technologies, the inventors have found that the existing playback frequency response equalization method has the problem of being easily affected by interference noise in the environment and having poor robustness. Summary of the invention

[0005] In order to improve measurement accuracy and enhance robustness, the present application provides a playback frequency response adaptive equalization method and device.

[0006] In a first aspect, the present application provides a playback frequency response adaptive equalization method.

[0007] This application is achieved through the following technical solutions:

[0008] A playback frequency response adaptive equalization method comprises the following steps:

[0009] Obtain the original playback signal of the speaker and the audio signal collected by the microphone;

[0010] Designing a room transfer function between the original playback signal and the audio signal by using a preset adaptive filtering algorithm;

[0011] estimating a frequency response compensation curve according to the room transfer function;

[0012] Based on the frequency response compensation curve, a corresponding IIR filter is designed to perform frequency response compensation on the original playback signal.

[0013] In a preferred example, the present application may be further configured as follows: the step of designing the room transfer function between the original playback signal and the audio signal by using a preset adaptive filtering algorithm includes:

[0014] Based on the original playback signal and the audio signal, the ambient audio in the audio signal is made equal to a preset input signal, and a filter response is calculated;

[0015] When the preset minimum error condition is met, the filter response is output, and the obtained filter response is used as the room transfer function between the original playback signal and the audio signal.

[0016] In a preferred example, the present application can be further configured as follows: it also includes the following steps:

[0017] When the error of y(n)r(n)-u(n) is the smallest, the minimum error condition is met, wherein y(n) is the original playback signal of the speaker, r(n) is the filter response, and u(n) is the preset input signal.

[0018] In a preferred example, the present application may be further configured as follows: according to the room transfer function, the step of estimating the frequency response compensation curve includes:

[0019] Perform fast Fourier transform based on the filter response to obtain the frequency response of the filter;

[0020] Modulo the frequency response to obtain an amplitude-frequency response of the filter;

[0021] A target amplitude-frequency response complementary to the amplitude-frequency response is solved to obtain the frequency response compensation curve.

[0022] In a preferred example, the present application may be further configured as follows: based on the frequency response compensation curve, the step of designing a corresponding IIR filter to perform frequency response compensation on the original playback signal includes:

[0023] Calculating power spectrum density using the frequency response compensation curve;

[0024] Performing an inverse Fourier transform on the power spectral density to obtain an autocorrelation function;

[0025] According to the autocorrelation function, solving the filter a coefficient;

[0026] An input matrix is ​​constructed based on the ambient audio in the audio signal, and an output matrix is ​​constructed based on the output signal of the IIR filter to solve the filter b coefficient.

[0027] In a preferred example, the present application can be further configured as follows: according to the autocorrelation function, the step of solving the filter a coefficient includes:

[0028] Based on the autocorrelation function and the filter a coefficient, combined with a preset order, an autoregressive model is constructed;

[0029] The autoregressive model is solved by linear algebra method using the Yule-Walker equation to obtain the filter a coefficient.

[0030] In a preferred example, the present application can be further configured as follows: it also includes the following steps:

[0031] Compensating the overall gain of the IIR filter to obtain a compensated IIR filter;

[0032] The compensated IIR filter is applied to the original playback signal to perform frequency response compensation.

[0033] In a preferred example, the present application may be further configured as follows: the overall gain of the IIR filter is compensated, and the steps of obtaining the compensated IIR filter include:

[0034] Generate a white noise sequence of preset length;

[0035] Inputting the white noise sequence into the IIR filter to obtain a filtered signal;

[0036] Calculating the energy of the white noise sequence and the filtered signal respectively to determine a gain factor;

[0037] The filter b coefficient is adjusted according to the gain factor to obtain a compensated IIR filter.

[0038] In a second aspect, the present application provides a playback frequency response adaptive equalization device.

[0039] This application is achieved through the following technical solutions:

[0040] A playback frequency response adaptive equalization device, comprising:

[0041] The acquisition module is used to obtain the original playback signal of the speaker and the audio signal collected by the microphone;

[0042] An adaptive module, used to design a room transfer function between the original playback signal and the audio signal by using a preset adaptive filtering algorithm;

[0043] A frequency response compensation module, used for estimating a frequency response compensation curve according to the room transfer function;

[0044] The frequency response equalization module is used to design a corresponding IIR filter based on the frequency response compensation curve to perform frequency response compensation on the original playback signal.

[0045] In a third aspect, the present application provides a computer device.

[0046] This application is achieved through the following technical solutions:

[0047] A computer device comprises a memory, a processor and a computer program stored in the memory and executable on the processor, wherein the processor implements the steps of any one of the above-mentioned playback frequency response adaptive equalization methods when executing the computer program.

[0048] In summary, compared with the prior art, the technical solution provided by this application has at least the following beneficial effects:

[0049] The original playback signal of the speaker and the audio signal collected by the microphone are obtained for estimating the acoustic transfer function of the speaker; the room transfer function between the original playback signal and the audio signal is designed by using a preset adaptive filtering algorithm, and the acoustic transfer change relationship including the signal amplitude and phase from the original playback signal to the microphone after passing through the speaker and the acoustic environment is adaptively obtained, thereby reducing the influence of interference noise in the environment, and realizing the measurement of the frequency response of the speaker system in a noisy environment, which is conducive to accurately estimating the frequency response in a noisy environment, the measurement accuracy is improved, and the calculation complexity is reduced compared with the traditional method; according to the room transfer function, the frequency response compensation curve is estimated for frequency response compensation and playback frequency response equalization; based on the frequency response compensation curve, the corresponding IIR filter is designed to compensate the frequency response of the original playback signal, which can realize complex frequency response with fewer filter coefficients, is more suitable for real-time audio processing applications, improves the measurement accuracy of the frequency response, and has a simple measurement method, enhanced robustness, and significantly improved the frequency response performance of the speaker system. BRIEF DESCRIPTION OF THE DRAWINGS

[0050] Figure 1 A schematic diagram of the main flow of a playback frequency response adaptive equalization method provided by an exemplary embodiment of the present application.

[0051] Figure 2 A schematic diagram of estimating a room transfer function of a playback frequency response adaptive equalization method provided by yet another exemplary embodiment of the present application.

[0052] Figure 3 A structural block diagram of a playback frequency response adaptive equalization device provided for another exemplary embodiment of the present application.

[0053] Figure 4 A system implementation block diagram of a playback frequency response adaptive equalization method provided for an exemplary embodiment of the present application. DETAILED DESCRIPTION

[0054] This specific embodiment is merely an explanation of the present application and is not a limitation of the present application. After reading this specification, those skilled in the art may make modifications to the present embodiment without any creative contribution as needed, but such modifications are protected by the patent law as long as they are within the scope of the claims of the present application.

[0055] In order to make the purpose, technical solution and advantages of the embodiments of the present application clearer, the technical solution in the embodiments of the present application will be clearly and completely described below in conjunction with the drawings in the embodiments of the present application. Obviously, the described embodiments are part of the embodiments of the present application, not all of the embodiments. Based on the embodiments in the present application, all other embodiments obtained by ordinary technicians in this field without creative work are within the scope of protection of this application.

[0056] In addition, the term "and / or" in this article is only a description of the association relationship of associated objects, indicating that there can be three relationships. For example, A and / or B can represent: A exists alone, A and B exist at the same time, and B exists alone. In addition, the character " / " in this article, unless otherwise specified, generally means that the associated objects before and after are in an "or" relationship.

[0057] The embodiments of the present application are further described in detail below in conjunction with the drawings in the specification.

[0058] Reference Figure 1 , an embodiment of the present application provides a playback frequency response adaptive equalization method, and the main steps of the method are described as follows.

[0059] S1: Obtain the original playback signal of the speaker and the audio signal collected by the microphone;

[0060] S2: Designing a room transfer function between the original playback signal and the audio signal by using a preset adaptive filtering algorithm; wherein the adaptive filtering algorithm may adopt an LMS (Least Mean Squares) algorithm, an RLS (Recursive Least Squares) algorithm or an NLMS (Normalized Least Mean Squares) algorithm to extract the frequency response characteristics of the loudspeaker, and has good convergence and stability;

[0061] S3: estimating a frequency response compensation curve according to the room transfer function;

[0062] S4: Based on the frequency response compensation curve, design a corresponding IIR filter to perform frequency response compensation on the original playback signal.

[0063] In one embodiment, the step of designing a room transfer function between the original playback signal and the audio signal using a preset adaptive filtering algorithm includes:

[0064] Based on the original playback signal and the audio signal, the ambient audio in the audio signal is made equal to a preset input signal, and a filter response is calculated;

[0065] When the preset minimum error condition is met, the filter response is output, and the obtained filter response is used as the room transfer function between the original playback signal and the audio signal.

[0066] Reference Figure 2 Specifically, the digital signal played by the system through the speaker is y(n), that is, the original playback signal. The audio signal collected by the microphone is s(n)=x(n)+r(n), where x(n) is various sounds in the environment, including noise, human voice, etc., and r(n) is the signal picked up by the microphone after the speaker sound is acoustically transmitted.

[0067] Input s(n) and y(n) into the LMS adaptive filter, where y(n) is the desired signal of the adaptive filter and u(n) is the input signal of the adaptive filter, so that u(n)=x(n) in the system, and calculate the filter response r(n). The filter response r(n) reflects the acoustic transmission relationship of the original playback signal from the speaker and the acoustic environment to the microphone. The filter response r(n) includes the change relationship between the amplitude and phase of the signal.

[0068] When the preset minimum error condition is met, the filter response is output as a room transfer function between the original playback signal and the audio signal.

[0069] This embodiment uses a preset LMS algorithm to adaptively obtain the room transfer function between the original playback signal of the speaker and the microphone collection signal, thereby realizing an adaptive filter function. Compared with traditional methods, this method can accurately estimate the frequency response in a noisy environment with lower computational complexity.

[0070] In one embodiment, a playback frequency response adaptive equalization method further includes the following steps:

[0071] When the error of y(n)r(n)-u(n) is the smallest, the minimum error condition is met, wherein y(n) is the original playback signal of the speaker, r(n) is the filter response, and u(n) is the preset input signal.

[0072] By calculating the filter response r(n), the error of y(n)r(n)-u(n) is minimized, which is conducive to accurately estimating the frequency response in a noisy environment and improving the measurement accuracy. At this time, r(n) is a finite impulse response (FIR) filter.

[0073] Next, the frequency response is calculated based on the room transfer function to obtain the frequency response change of the original playback signal transmitted to the microphone. The amplitude-frequency response is then calculated to obtain the power spectrum amplitude change of the audio signal collected by the microphone at different frequencies relative to the original playback signal, which is used to estimate the frequency response compensation curve.

[0074] In one embodiment, the step of estimating a frequency response compensation curve according to the room transfer function comprises:

[0075] Perform fast Fourier transform based on the filter response to obtain the frequency response of the filter;

[0076] Modulo the frequency response to obtain an amplitude-frequency response of the filter;

[0077] A target amplitude-frequency response complementary to the amplitude-frequency response is solved to obtain the frequency response compensation curve.

[0078] By performing FFT transformation on the filter response r(n), the frequency response of the filter is obtained. The specific formula is as follows:

[0079] H(f)=FFT(r(n))

[0080] Where H(f) is the frequency response of the filter, f=1,2…,N, and N is the number of points in the frequency domain.

[0081] The frequency response H(f) of the filter is modulo to obtain the amplitude-frequency response |H(f)| of the filter.

[0082] The goal of frequency response compensation is to make the amplitude response at each frequency point the same, so the target amplitude-frequency response |H(f)| is solved to be complementary to the amplitude-frequency response |H(f)|. comp (f)|, the sum of the two is the flat curve |H(f)|+|H comp (f)|=1, then |H comp (f)|=1-|H(f)| is the frequency response compensation curve.

[0083] By calculating the frequency response of the room transfer function, a frequency response curve is obtained, and a compensation curve complementary to the frequency response of the room transfer function is designed to achieve frequency response adaptive equalization.

[0084] In one embodiment, a machine learning model may also be used to predict and compensate for frequency response characteristics by training a neural network, and output a frequency response compensation curve to achieve frequency response adaptive equalization of the speaker system.

[0085] Finally, based on the frequency response compensation curve, the corresponding IIR (Infinite Impulse Response) filter is designed to perform frequency response compensation on the original playback signal, so as to achieve efficient frequency characteristics with a lower order.

[0086] In one embodiment, based on the frequency response compensation curve, the step of designing a corresponding IIR filter to perform frequency response compensation on the original playback signal includes:

[0087] Calculating power spectrum density using the frequency response compensation curve;

[0088] Performing an inverse Fourier transform on the power spectral density to obtain an autocorrelation function;

[0089] According to the autocorrelation function, solving the filter a coefficient;

[0090] An input matrix is ​​constructed based on the ambient audio in the audio signal, and an output matrix is ​​constructed based on the output signal of the IIR filter to solve the filter b coefficient.

[0091] Specifically, when obtaining the frequency response compensation curve |H comp (f)|, an IIR filter with a specific amplitude-frequency response is designed, where the order of the IIR filter is n and its differential equation is:

[0092] y(n)+a 1 y(n-1)+…+a n y(0)=b 0 x(n)+b 1 x(n-1)+…+b n x(0)

[0093] In the formula, a i and b i are the coefficients of the IIR filter, x(n) is the input signal of the IIR filter, and y(n) is the output signal of the IIR filter.

[0094] The steps to design an IIR filter are as follows:

[0095] First, the target amplitude-frequency response is defined using the frequency response compensation curve, that is, H mag (f)=|H comp (f)|;

[0096] Next, the power spectrum density is calculated based on the target amplitude-frequency response, that is, P(f) = |H mag(f)| 2 ;

[0097] Secondly, the power spectrum density is transformed by IFFT to obtain the autocorrelation function, that is, R(k)=IFFT(P(f)), where k is the lag coefficient of the autocorrelation function;

[0098] Finally, according to the autocorrelation function, the filter a coefficient is solved, that is, a i ; and construct an input matrix based on the ambient audio in the audio signal, construct an output matrix based on the output signal of the IIR filter, and solve the filter b coefficient, that is, b i .

[0099] In one embodiment, the step of solving the filter a coefficient according to the autocorrelation function includes:

[0100] Based on the autocorrelation function and the filter a coefficient, combined with a preset order, an autoregressive model is constructed;

[0101] The autoregressive model is solved by linear algebra method using the Yule-Walker equation to obtain the filter a coefficient.

[0102] The parameters of the autoregressive (AR) model are solved by the Yule-Walker equation. Specifically, an AR model of order p is defined, and the Yule-Walker equation can be expressed as:

[0103] R(k)=a 1 R(k-1)+a 2 R(k-2)+…+a P R(kp)

[0104] Where R(k) is the known autocorrelation function and the filter coefficient a is i ,i=1,2,…,p, where p is the order of the AR model.

[0105] By solving the linear equation using standard linear algebra methods, the filter coefficient a can be obtained: i .

[0106] In one embodiment, an input matrix is ​​constructed based on the ambient audio in the audio signal, and an output matrix is ​​constructed based on the output signal of the IIR filter, and the filter b coefficient is solved, including:

[0107] Construct the input matrix X, which is represented as follows:

[0108]

[0109] Where x(n) is the ambient audio in the audio signal, N is the signal length, and different columns of the input matrix X are different delayed versions of the input signal x(n).

[0110] Construct the output matrix Y, which is expressed as follows:

[0111]

[0112] N is the signal length.

[0113] According to the input matrix and the output matrix, a linear equation system is constructed, that is, Xb=Y, and the filter b coefficient b is obtained by solving it through the least squares method (Recursive Least Squares, RLS). i .

[0114] Based on the frequency response compensation curve, by fitting the autocorrelation function of the target frequency response, solving the linear equations consisting of the Yule-Walker equation and the input-output matrix, the feedback coefficient and feedforward coefficient of the IIR filter are obtained, and then the filter coefficient a can be used. i and filter b coefficient b i The constructed IIR filter performs frequency response compensation on the original playback signal and can achieve complex frequency response with fewer filter coefficients, which is more suitable for real-time audio processing applications.

[0115] In one embodiment, a FIR (Finite Impulse Response) filter may also be used to perform frequency response compensation on the original playback signal, and more coefficients may be designed to have better phase response characteristics, which will not be described in detail here.

[0116] Since the obtained filter amplitude-frequency response is the amplitude relationship between the microphone acquisition signal and the original playback signal, it includes the overall volume attenuation on the acoustic propagation path and the influence of the microphone sensitivity. Therefore, in order to ensure that the loudspeaker playback volume before and after compensation is relatively consistent, in one embodiment, a playback frequency response adaptive equalization method further includes the following steps:

[0117] Compensating the overall gain of the IIR filter to obtain a compensated IIR filter;

[0118] The compensated IIR filter is applied to the original playback signal to perform frequency response compensation, so as to additionally compensate for the overall gain of the filter.

[0119] In one embodiment, the step of compensating the overall gain of the IIR filter to obtain a compensated IIR filter includes:

[0120] Generate a white noise sequence of preset length;

[0121] Inputting the white noise sequence into the IIR filter to obtain a filtered signal;

[0122] Calculating the energy of the white noise sequence and the filtered signal respectively to determine a gain factor;

[0123] The filter b coefficient is adjusted according to the gain factor to obtain a compensated IIR filter.

[0124] Specifically, x(n)=randn(N,1) is used to generate a white noise sequence x(n) with a length of N.

[0125] Next, apply the IIR filter and pass the white noise signal x(n) through the IIR filter to obtain the filtered signal y(n), that is,

[0126] Where b k is the numerator coefficient of the IIR filter, which represents the weighted sum of the input signal to control the direct impact of the input signal on the current output; q represents the order of the numerator polynomial of the IIR filter actually used; a k is the denominator coefficient of the IIR filter, which represents the weighted sum of the historical output signals to control the influence of the historical output signals on the current output signal; p represents the order of the denominator polynomial of the IIR filter actually used.

[0127] Then calculate the energy of the input signal x(n) and the output signal y(n) respectively to determine the gain factor G, where the energy of the input signal x(n) is:

[0128]

[0129] The energy of the output signal y(n) is:

[0130]

[0131] The gain factor is then:

[0132]

[0133] Apply the gain factor G to the IIR filter coefficients to obtain the new filter coefficients:

[0134] a=[1,a 1 ,a 2 ,...,a p ]

[0135] b G =G×[b 0 ,b 1 ,...,b q ]

[0136] Finally, the IIR filter after compensation gain is used to perform frequency response equalization compensation on the original playback signal to ensure that the volume of the speaker playback is relatively consistent before and after compensation.

[0137] In one embodiment, the dynamic range compression technology or the automatic gain control (AGC) technology may also be used to compensate the overall gain of the IIR filter to achieve consistency in the speaker playback volume before and after compensation.

[0138] In summary, a playback frequency response adaptive equalization method obtains the original playback signal of the speaker and the audio signal collected by the microphone for the estimation of the acoustic transfer function of the speaker; uses the preset LMS algorithm to design the room transfer function between the original playback signal and the audio signal, and adaptively obtains the acoustic transfer change relationship including signal amplitude and phase from the original playback signal to the microphone after passing through the speaker and the acoustic environment, thereby reducing the influence of interference noise in the environment, and realizing the measurement of the speaker system frequency response in a noisy environment, which is conducive to accurately estimating the frequency response in a noisy environment, and the measurement accuracy is improved, and the calculation complexity is reduced compared with the traditional method; according to the room transfer function, the frequency response compensation curve is estimated for frequency response compensation and playback frequency response equalization; based on the frequency response compensation curve, the corresponding IIR filter is designed to compensate the frequency response of the original playback signal, which can realize complex frequency response through fewer filter coefficients, is more suitable for real-time audio processing applications, improves the measurement accuracy of the frequency response, and has a simple measurement method, enhanced robustness, and significantly improves the frequency response performance of the speaker system.

[0139] A playback frequency response adaptive equalization method can accurately estimate the frequency response characteristics of the speaker in a noisy environment by introducing an adaptive filtering LMS algorithm, solving the problem of high requirements for test signals in traditional methods, reducing the impact of environmental noise on measurement results, and improving the accuracy of frequency response equalization. At the same time, compared with the traditional impulse response decomposition method, the adaptive filtering LMS algorithm has lower computational complexity and is suitable for application in real-time systems, reducing dependence on professional equipment and technology and reducing computational complexity. In addition, the adaptive filter can be dynamically adjusted in different environments, has strong adaptability, has high system robustness, and can work stably in various complex environments.

[0140] A playback frequency response adaptive equalization method realizes complex frequency response with fewer filter coefficients by introducing IIR filter, which is suitable for real-time audio processing applications and ensures the real-time performance and response speed of the system.

[0141] A playback frequency response adaptive equalization method is based on an automatic equalization algorithm for playback frequency response based on an adaptive filter. It can accurately measure and compensate the frequency response of a speaker system in a noisy environment. The measurement method is simple, and the frequency response calculation and compensation can be automatically completed on a real-time system without the need for professional equipment for measurement. The requirements for test signals are low, and compensation can be achieved through real-time convergence and calculation during a call. The method can be applied to real-time conference systems, and the system is more versatile.

[0142] It should be understood that the size of the serial numbers of the steps in the above embodiments does not mean the order of execution. The execution order of each process should be determined by its function and internal logic, and should not constitute any limitation on the implementation process of the embodiments of the present application.

[0143] Reference Figure 3 The embodiment of the present application also provides a playback frequency response adaptive equalization device, which corresponds one-to-one to a playback frequency response adaptive equalization method in the above embodiment. The playback frequency response adaptive equalization device includes:

[0144] The acquisition module is used to obtain the original playback signal of the speaker and the audio signal collected by the microphone;

[0145] An adaptive module, used to design a room transfer function between the original playback signal and the audio signal by using a preset adaptive filtering algorithm;

[0146] A frequency response compensation module, used for estimating a frequency response compensation curve according to the room transfer function;

[0147] The frequency response equalization module is used to design a corresponding IIR filter based on the frequency response compensation curve to perform frequency response compensation on the original playback signal.

[0148] A playback frequency response adaptive equalization device also includes:

[0149] The gain compensation module is used to compensate the overall gain of the IIR filter to obtain a compensated IIR filter, and apply the compensated IIR filter to the original playback signal for frequency response compensation.

[0150] Reference Figure 4 Specifically, the signal processor obtains the original playback signal of the speaker and the audio signal collected by the microphone, and adaptively obtains the room transfer function between the original playback signal and the audio signal through a preset adaptive filtering algorithm.

[0151] The signal processor calculates the energy of the room transfer function to obtain a playback frequency response curve, then compensates the playback frequency response curve to make it flat to obtain an amplitude compensation curve, and estimates the frequency response compensation curve.

[0152] Finally, based on the frequency response compensation curve, the signal processor designs a corresponding IIR filter to perform frequency response compensation on the original playback signal to achieve frequency response equalization.

[0153] For the specific definition of a playback frequency response adaptive equalization device, please refer to the definition of a playback frequency response adaptive equalization method mentioned above, which will not be repeated here.

[0154] Each module in the above-mentioned playback frequency response adaptive equalization device can be implemented in whole or in part by software, hardware, or a combination thereof. Each of the above-mentioned modules can be embedded in or independent of a processor in a computer device in the form of hardware, or can be stored in a memory in a computer device in the form of software, so that the processor can call and execute the operations corresponding to each of the above modules.

[0155] In one embodiment, a computer device is provided, which may be a server. The computer device includes a processor, a memory, a network interface, and a database connected via a system bus. The processor of the computer device is used to provide computing and control capabilities. The memory of the computer device includes a non-volatile storage medium and an internal memory. The non-volatile storage medium stores an operating system, a computer program, and a database. The internal memory provides an environment for the operation of the operating system and the computer program in the non-volatile storage medium. The network interface of the computer device is used to communicate with an external terminal via a network connection. When the computer program is executed by the processor, any one of the above-mentioned playback frequency response adaptive equalization methods is implemented.

[0156] In one embodiment, a computer-readable storage medium is provided, including a memory, a processor, and a computer program stored in the memory and executable on the processor, and when the processor executes the computer program, any one of the above-mentioned playback frequency response adaptive equalization methods is implemented.

[0157] In one embodiment, a computer program product is provided. The computer program product includes a computer program. When the computer program is executed by a processor, any one of the above-mentioned playback frequency response adaptive equalization methods is implemented.

[0158] Those of ordinary skill in the art can understand that all or part of the processes in the above-mentioned embodiments can be completed by instructing the relevant hardware through a computer program, and the computer program can be stored in a non-volatile computer-readable storage medium, including several instructions to enable a computer device (which can be a personal computer, a server, or a network device, etc.) to perform all or part of the steps of the methods described in each embodiment of the present application. When the computer program is executed, it may include the processes of the embodiments of the above-mentioned methods. Among them, any reference to memory, storage, database or other media used in the embodiments provided in the present application may include non-volatile and / or volatile memory. Non-volatile memory may include read-only memory (ROM), programmable ROM (PROM), electrically programmable ROM (EPROM), electrically erasable programmable ROM (EEPROM) or flash memory. Volatile memory may include random access memory (RAM) or external cache memory. By way of illustration and not limitation, RAM is available in many forms, such as static RAM (SRAM), dynamic RAM (DRAM), synchronous DRAM (SDRAM), double data rate SDRAM (DDRSDRAM), enhanced SDRAM (ESDRAM), synchronous link (Synchlink) DRAM (SLDRAM), memory bus (Rambus) direct RAM (RDRAM), direct memory bus dynamic RAM (DRDRAM), and memory bus dynamic RAM (RDRAM), etc.

[0159] Those skilled in the art will clearly understand that for the sake of convenience and brevity of description, only the division of the above-mentioned functional units and modules is used as an example. In actual applications, the above-mentioned functions can be distributed and completed by different functional units and modules as needed, that is, the internal structure of the system can be divided into different functional units or modules to complete all or part of the functions described above.

Claims

1. A playback frequency response adaptive equalization method, characterized in that: The following steps are included: Obtain the original playback signal of the speaker and the audio signal collected by the microphone; Designing a room transfer function between the original playback signal and the audio signal by using a preset adaptive filtering algorithm; estimating a frequency response compensation curve according to the room transfer function; Based on the frequency response compensation curve, a corresponding IIR filter is designed to perform frequency response compensation on the original playback signal.

2. The playback frequency response adaptive equalization method according to claim 1, characterized in that: The step of designing the room transfer function between the original playback signal and the audio signal by using a preset adaptive filtering algorithm comprises: Using an LMS algorithm, based on the original playback signal and the audio signal, making the ambient audio in the audio signal equal to a preset input signal, and calculating a filter response; When the preset minimum error condition is met, the filter response is output, and the obtained filter response is used as the room transfer function between the original playback signal and the audio signal.

3. The playback frequency response adaptive equalization method according to claim 2, characterized in that: The following steps are also included: When the error of y(n)r(n)-u(n) is the smallest, the minimum error condition is met, wherein y(n) is the original playback signal of the speaker, r(n) is the filter response, and u(n) is the preset input signal.

4. The playback frequency response adaptive equalization method according to claim 1, characterized in that: According to the room transfer function, the step of estimating the frequency response compensation curve comprises: Perform fast Fourier transform based on the filter response to obtain the frequency response of the filter; Modulo the frequency response to obtain an amplitude-frequency response of the filter; A target amplitude-frequency response complementary to the amplitude-frequency response is solved to obtain the frequency response compensation curve.

5. The playback frequency response adaptive equalization method according to claim 1, characterized in that: Based on the frequency response compensation curve, the step of designing a corresponding IIR filter to perform frequency response compensation on the original playback signal includes: Calculating power spectrum density using the frequency response compensation curve; Performing an inverse Fourier transform on the power spectral density to obtain an autocorrelation function; According to the autocorrelation function, solving the filter a coefficient; An input matrix is ​​constructed based on the ambient audio in the audio signal, and an output matrix is ​​constructed based on the output signal of the IIR filter to solve the filter b coefficient.

6. The playback frequency response adaptive equalization method according to claim 5, characterized in that: According to the autocorrelation function, the step of solving the filter a coefficient includes: Based on the autocorrelation function and the filter a coefficient, combined with a preset order, an autoregressive model is constructed; The autoregressive model is solved by linear algebra method using the Yule-Walker equation to obtain the filter a coefficient.

7. The playback frequency response adaptive equalization method according to any one of claims 1 to 6, characterized in that: The following steps are also included: Compensating the overall gain of the IIR filter to obtain a compensated IIR filter; The compensated IIR filter is applied to the original playback signal to perform frequency response compensation.

8. The playback frequency response adaptive equalization method according to claim 7, characterized in that: The step of compensating the overall gain of the IIR filter to obtain a compensated IIR filter comprises: Generate a white noise sequence of preset length; Inputting the white noise sequence into the IIR filter to obtain a filtered signal; Calculating the energy of the white noise sequence and the filtered signal respectively to determine a gain factor; The filter b coefficient is adjusted according to the gain factor to obtain a compensated IIR filter.

9. A playback frequency response adaptive equalization device, characterized in that: include, The acquisition module is used to obtain the original playback signal of the speaker and the audio signal collected by the microphone; An adaptive module, used to design a room transfer function between the original playback signal and the audio signal by using a preset adaptive filtering algorithm; A frequency response compensation module, used for estimating a frequency response compensation curve according to the room transfer function; The frequency response equalization module is used to design a corresponding IIR filter based on the frequency response compensation curve to perform frequency response compensation on the original playback signal.

10. A computer device, characterized in that: The method comprises a memory, a processor and a computer program stored in the memory, wherein the processor executes the computer program to implement the steps of the method according to any one of claims 1 to 8.

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