A precise compensation method for an acoustic signal simulator

Abstract

This invention discloses a precise compensation method for an acoustic signal simulator. The compensation method includes the following steps: First, generating multiple sets of single-frequency digital signals with different amplitudes and frequencies; Second, passing the generated digital signals through an acoustic signal simulator to obtain output acoustic signals; Third, measuring the sound level of the output acoustic signals using a sound level meter to obtain the correspondence between sound level, digital amplitude, and frequency; Fourth, for the final desired acoustic signal, generating its corresponding digital signal time-domain waveform, dividing it into frames, and calculating the power spectrum of each frame; Fifth, estimating the gain at different frequency points of each frame signal based on the correspondence between sound level, digital amplitude, and frequency; Sixth, calculating the frequency response of the compensation filter corresponding to each frame of digital signal and filtering the digital signal; Seventh, merging the filtered digital signals of each frame and using them to drive the acoustic signal simulator to obtain a precise acoustic signal.

Description

Technical Field

[0001] This invention belongs to the field of signal processing, and in particular relates to a precise compensation method for an acoustic signal simulator. Background Technology

[0002] In the process of achieving high-precision signal reconstruction, acoustic signal simulators often introduce complex nonlinear distortions in their digital-to-analog conversion, power amplification, and transducer stages. These distortions not only vary with the signal frequency, but more importantly, at the same frequency, the system gain changes significantly with the dynamic changes in the input signal amplitude, exhibiting strong coupling characteristics. Therefore, the system output is not a linear superposition of the inputs and cannot be accurately predicted and compensated for using traditional frequency-domain linear models or global correction with fixed coefficients.

[0003] Existing compensation methods typically do not consider the time-varying characteristics of the signal, assuming it to be stationary. Conventional methods usually rely on statistical analysis of the entire signal, neglecting the time-varying effects of transient components or local spectral features, resulting in insufficient compensation for short-lived frequency components and significant limitations in instantaneous response. Furthermore, the designed response is usually fixed, making it difficult to adapt to the dynamic nonlinear behavior of the system.

[0004] Therefore, there is an urgent need for a dynamic compensation mechanism that can identify and adapt to the combined nonlinear changes in amplitude and frequency in real time, while also having the ability to perform high-resolution analysis and processing of the local spectral characteristics of the signal, so as to improve the overall output accuracy and adaptability of the acoustic signal simulation system. Summary of the Invention

[0005] Objective: To address the above-mentioned problems, this invention proposes a precise compensation method for an acoustic signal simulator. This method uses interpolation techniques to refine the channel information based on collected channel characteristic data, and then compensates the generated digital signal to effectively overcome nonlinear effects in the system. Through this method, we can simulate and compensate for the nonlinear characteristics of a digital microphone, thereby significantly improving the accuracy and reliability of the generated signal. This invention's precise acoustic signal generation method for a nonlinear response acoustic signal simulator comprehensively considers the influence of the acoustic signal simulator's characteristics on the output signal, providing strong technical support for achieving efficient and stable signal generation.

[0006] Technical Solution: To achieve the above objectives, this invention proposes a precise compensation method for an acoustic signal simulator, which includes the following steps:

[0007] The first step is to generate multiple sets of single-frequency digital signals with different amplitudes and frequencies;

[0008] The second step is to use the generated digital signals to obtain the output sound signals through a sound signal simulator;

[0009] The third step is to use a sound level meter to measure the sound level of the output sound signal to obtain the correspondence between sound level, digital amplitude and frequency.

[0010] The fourth step is to generate the corresponding digital signal time-domain waveform for the final desired acoustic signal, divide it into frames, and calculate the power spectrum of each frame.

[0011] The fifth step is to estimate the gain at different frequency points of each frame of signal based on the correspondence between sound level, digital amplitude, and frequency.

[0012] The sixth step is to calculate the frequency response of the compensation filter corresponding to each frame of digital signal and filter the digital signal.

[0013] The seventh step is to merge each frame of the filtered digital signal and use it to drive the sound signal simulator to obtain an accurate sound signal.

[0014] Furthermore, in step one, multiple sets of single-frequency digital signals with different amplitudes and frequencies are generated, as detailed below:

[0015] (2.1) Set multiple non-repeating numerical amplitudes ,and ,in, The number of numerical amplitudes. It is the minimum input amplitude of the sound signal simulator. This is the maximum input amplitude of the sound signal simulator, allowing you to set multiple non-repeating frequencies. ,and ,in, The number of frequencies. It is the minimum input frequency of the sound signal simulator. This is the maximum input frequency of the acoustic signal simulator;

[0016] (2.2) Generate multiple sets of single-frequency signals according to the determined parameters. :

[0017]

[0018] in, The digital amplitude of the signal. , For the frequency of the signal, , The time of the signal.

[0019] Furthermore, in step two, the generated multiple sets of single-frequency digital signals are respectively used to obtain the output sound signals through an acoustic signal simulator. .

[0020] Furthermore, in step three, the output signal is measured using a sound level meter. sound level , obtain sound level and numerical amplitude Single-frequency signal frequency The correspondence, where, The digital amplitude of the signal. , For the frequency of the signal, , This represents the sound level of the output signal from the nonlinear acoustic signal simulator, measured in dB.

[0021] Furthermore, in step four, for the final desired acoustic signal, its corresponding digital signal time-domain waveform is generated. , , Represents time-domain waveform The points will Divided into Frame, each frame is denoted as , , This represents the number of points in a frame and calculates the corresponding power spectrum. , Power spectrum The calculation method is as follows:

[0022]

[0023] in, Represents the modulo operation This represents the floor operation. It is the imaginary unit.

[0024] Furthermore, in step five, based on the correspondence between sound level, digital amplitude, and frequency, the gain at different frequency points of each frame of signal is estimated, as follows:

[0025] (6.1) Calculate the power spectrum Corresponding frequency vector :

[0026]

[0027] in, The sampling frequency of the acoustic signal simulator;

[0028] (6.2) For sets elements in If satisfied and , making Record the conditions that are met. The set is Further calculate the corresponding gain :

[0029]

[0030] (6.3) Record those that do not meet the conditions. The set is ,but For sets elements in The gain calculation method is as follows:

[0031] First, regarding sets elements in ,remember Mid-range The three most recent elements are For sets Each element in The sound level is obtained by interpolation. exist The estimated value at the location :

[0032]

[0033] remember Mid-range The three most recent elements are The corresponding numerical range for ,by Fitting a curve for the independent variable:

[0034]

[0035] Let expression (6) equal to Calculate the corresponding numerical range ,calculate Corresponding gain:

[0036] .

[0037] Furthermore, in step six, the frequency response of the compensation filter corresponding to each frame of the digital signal is calculated, and the digital signal is filtered as follows:

[0038] (7.1) Calculate the magnitude of the gain of the compensation filter. :

[0039]

[0040] in, This represents the integer division operation;

[0041] (7.2) Calculate the phase of the filter gain :

[0042]

[0043] in, The set group delay constant, The imaginary unit, The number of points in the time-domain waveform;

[0044] (7.3) The frequency response of the compensation filter is calculated. :

[0045]

[0046] (7.4) For signals Compensation filtering is performed to obtain the filtered signal. :

[0047]

[0048] in, , This represents the number of points in each frame of the time-domain waveform. The imaginary unit, To compensate for the frequency response of the filter.

[0049] Furthermore, in step seven, each frame of the filtered digital signal is merged. For overlapping parts, the signals are merged by averaging, while non-overlapping parts are merged directly. The merged signal is then used to drive the acoustic signal simulator to obtain an accurate acoustic signal.

[0050] Beneficial effects: Compared with existing technologies, the technical solution of the present invention has the following beneficial technical effects:

[0051] Existing compensation methods typically fail to consider the time-varying characteristics of the signal, assuming it to be stationary. The designed responses are usually fixed and difficult to adapt to dynamic nonlinear behavior. Conventional methods often rely on statistical analysis of the entire signal, neglecting the time-varying effects of transient components or local spectral features, leading to insufficient compensation for short-lived frequency components and significant limitations in instantaneous response. Therefore, this method fully considers the nonlinear characteristics of the system, estimating the gain at different frequency points based on collected channel characteristic data. This effectively overcomes the nonlinear effects of the acoustic signal simulator. Furthermore, considering the non-stationarity of the signal, the compensated signal better meets the requirements of the actual system, providing a more effective solution for channel spectrum compensation in acoustic signal simulators. Attached Figure Description

[0052] Figure 1 This is a flowchart of the present invention;

[0053] Figure 2 The time-domain waveform of an ideal signal;

[0054] Figure 3 The time spectrum of an ideal signal;

[0055] Figure 4 The time-domain waveform of the output signal;

[0056] Figure 5 This represents the time spectrum of the output signal. Detailed Implementation

[0057] The technical solutions of the present invention will be clearly and completely described below with reference to the accompanying drawings of the embodiments of the present invention. Obviously, the described embodiments are only some embodiments of the present invention, and not all embodiments. Based on the embodiments of the present invention, all other embodiments obtained by those skilled in the art without creative effort are within the scope of protection of the present invention.

[0058] like Figure 1 As shown, this invention proposes a precise compensation method for an acoustic signal simulator, which includes the following steps:

[0059] The first step is to generate multiple sets of single-frequency digital signals with different amplitudes and frequencies;

[0060] The second step is to use the generated digital signals to obtain the output sound signals through a sound signal simulator;

[0061] The third step is to use a sound level meter to measure the sound level of the output sound signal to obtain the correspondence between sound level, digital amplitude and frequency.

[0062] The fourth step is to generate the corresponding digital signal time-domain waveform for the final desired acoustic signal, divide it into frames, and calculate the power spectrum of each frame.

[0063] The fifth step is to estimate the gain at different frequency points of each frame of signal based on the correspondence between sound level, digital amplitude, and frequency.

[0064] The sixth step is to calculate the frequency response of the compensation filter corresponding to each frame of digital signal and filter the digital signal.

[0065] The seventh step is to merge each frame of the filtered digital signal and use it to drive the sound signal simulator to obtain an accurate sound signal.

[0066] Furthermore, in step one, multiple sets of single-frequency digital signals with different amplitudes and frequencies are generated, as detailed below:

[0067] (2.1) Set multiple non-repeating numerical amplitudes ,and ,in, The number of numerical amplitudes. It is the minimum input amplitude of the sound signal simulator. This is the maximum input amplitude of the sound signal simulator, allowing you to set multiple non-repeating frequencies. ,and ,in, The number of frequencies. It is the minimum input frequency of the sound signal simulator. This is the maximum input frequency of the acoustic signal simulator;

[0068] (2.2) Generate multiple sets of single-frequency signals according to the determined parameters. :

[0069]

[0070] in, The digital amplitude of the signal. , For the frequency of the signal, , The time of the signal.

[0071] Furthermore, in step two, the generated multiple sets of single-frequency digital signals are respectively used to obtain the output sound signals through an acoustic signal simulator. .

[0072] Furthermore, in step three, the output signal is measured using a sound level meter. sound level , obtain sound level and numerical amplitude Single-frequency signal frequency The correspondence, where, The digital amplitude of the signal. , For the frequency of the signal, , This represents the sound level of the output signal from the nonlinear acoustic signal simulator, measured in dB.

[0073] Furthermore, in step four, for the final desired acoustic signal, its corresponding digital signal time-domain waveform is generated. , , Represents time-domain waveform The points will Divided into Frame, each frame is denoted as , , This represents the number of points in a frame and calculates the corresponding power spectrum. , Power spectrum The calculation method is as follows:

[0074]

[0075] in, Represents the modulo operation This represents the floor operation. It is the imaginary unit.

[0076] Furthermore, in step five, based on the correspondence between sound level, digital amplitude, and frequency, the gain at different frequency points of each frame of signal is estimated, as follows:

[0077] (6.1) Calculate the power spectrum Corresponding frequency vector :

[0078]

[0079] in, The sampling frequency of the acoustic signal simulator;

[0080] (6.2) For sets elements in If satisfied and , making Record the conditions that are met. The set is Further calculate the corresponding gain :

[0081]

[0082] (6.3) Record those that do not meet the conditions. The set is ,but For sets elements in The gain calculation method is as follows:

[0083] First, regarding sets elements in ,remember Mid-range The three most recent elements are For sets Each element in The sound level is obtained by interpolation. exist The estimated value at the location :

[0084]

[0085] remember Mid-range The three most recent elements are The corresponding numerical range for ,by Fitting a curve for the independent variable:

[0086]

[0087] Let expression (6) equal to Calculate the corresponding numerical range ,calculate Corresponding gain:

[0088] .

[0089] Furthermore, in step six, the frequency response of the compensation filter corresponding to each frame of the digital signal is calculated, and the digital signal is filtered as follows:

[0090] (7.1) Calculate the magnitude of the gain of the compensation filter. :

[0091]

[0092] in, This represents the integer division operation;

[0093] (7.2) Calculate the phase of the filter gain :

[0094]

[0095] in, The set group delay constant, The imaginary unit, The number of points in the time-domain waveform;

[0096] (7.3) The frequency response of the compensation filter is calculated. :

[0097]

[0098] (7.4) For signals Compensation filtering is performed to obtain the filtered signal. :

[0099]

[0100] in, , This represents the number of points in each frame of the time-domain waveform. The imaginary unit, To compensate for the frequency response of the filter.

[0101] Furthermore, in step seven, each frame of the filtered digital signal is merged. For overlapping parts, the signals are merged by averaging, while non-overlapping parts are merged directly. The merged signal is then used to drive the acoustic signal simulator to obtain an accurate acoustic signal.

[0102] Example

[0103] The output sound level of a current sound signal simulator and numerical amplitude and frequency The correspondence is .

[0104] First, by setting , Multiple single-frequency signals are generated according to formula (1) and output through an acoustic signal simulator.

[0105] Secondly, the output sound level is measured using a sound level meter to obtain multiple sets of measured sound levels. .

[0106] Next, an ideal digital signal is generated, such as Figure 2 As shown, the power spectrum is obtained from equation (2), and the time spectrum is as follows. Figure 3 As shown.

[0107] Furthermore, the frequency vector is calculated using equation (3), and the gain is calculated using equations (4), (5), (6), and (7).

[0108] Next, the frequency response of the compensation filter is calculated using equations (8), (9), and (10).

[0109] Finally, the digital signal is passed through a compensation filter to adjust its frequency response before driving an acoustic signal simulator to obtain the final result. The time-domain waveform of the output acoustic signal is shown below. Figure 4 As shown, the time spectrum is as follows Figure 5 As shown.

Claims

1. A precise compensation method for an acoustic signal simulator, characterized in that, The method includes the following steps: The first step is to generate multiple sets of single-frequency digital signals with different amplitudes and frequencies; The second step is to use the generated digital signals to obtain the output sound signals through a sound signal simulator; The third step is to use a sound level meter to measure the sound level of the output sound signal to obtain the correspondence between sound level, digital amplitude and frequency. The fourth step is to generate the corresponding digital signal time-domain waveform for the final desired acoustic signal, divide it into frames, and calculate the power spectrum of each frame. The fifth step is to estimate the gain at different frequency points of each frame of signal based on the correspondence between sound level, digital amplitude, and frequency. The sixth step is to calculate the frequency response of the compensation filter corresponding to each frame of digital signal and filter the digital signal. The seventh step is to merge each frame of the filtered digital signal and use it to drive the sound signal simulator to obtain an accurate sound signal.

2. The precise compensation method for an acoustic signal simulator according to claim 1, characterized in that, In step one, multiple sets of single-frequency digital signals with different amplitudes and frequencies are generated. The specific method is as follows: (2.1) Set multiple non-repeating numerical amplitudes ,and ,in, The number of numerical amplitudes. It is the minimum input amplitude of the sound signal simulator. This is the maximum input amplitude of the sound signal simulator, allowing you to set multiple non-repeating frequencies. ,and ,in, The number of frequencies. It is the minimum input frequency of the sound signal simulator. This is the maximum input frequency of the acoustic signal simulator; (2.2) Generate multiple sets of single-frequency signals according to the determined parameters. : in, The digital amplitude of the signal. , For the frequency of the signal, , The time of the signal.

3. The precise compensation method for an acoustic signal simulator according to claim 2, characterized in that, In step two, the generated multiple sets of single-frequency digital signals are respectively passed through an acoustic signal simulator to obtain the output acoustic signals. .

4. The precise compensation method for an acoustic signal simulator according to claim 3, characterized in that, In step three, the output signal is measured using a sound level meter. sound level , obtain sound level and numerical amplitude Single-frequency signal frequency The correspondence, where, The digital amplitude of the signal. , For the frequency of the signal, , This represents the sound level of the output signal from the nonlinear acoustic signal simulator, measured in dB.

5. The precise compensation method for an acoustic signal simulator according to claim 4, characterized in that, In step four, for the desired final acoustic signal, its corresponding digital signal time-domain waveform is generated. , , Represents time-domain waveform The points will Divided into Frame, each frame is denoted as , , This represents the number of points in a frame and calculates the corresponding power spectrum. , Power spectrum The calculation method is as follows: in, Represents the modulo operation This represents the floor operation. It is the imaginary unit.

6. The precise compensation method for an acoustic signal simulator according to claim 5, characterized in that, In step five, based on the correspondence between sound level, digital amplitude, and frequency, the gain at different frequency points of each frame of signal is estimated, as follows: (6.1) Calculate the power spectrum Corresponding frequency vector : in, The sampling frequency of the acoustic signal simulator; (6.2) For sets elements in If satisfied and , making Record the conditions that are met. The set is Further calculate the corresponding gain : (6.3) Record those that do not meet the conditions. The set is ,but For sets elements in The gain calculation method is as follows: First, regarding sets elements in ,remember Mid-range The three most recent elements are For sets Each element in The sound level is obtained by interpolation. exist The estimated value at the location : remember Mid-range The three most recent elements are The corresponding numerical range for ,by Fitting a curve for the independent variable: Let expression (6) equal to Calculate the corresponding numerical range ,calculate Corresponding gain: 。 7. The precise compensation method for an acoustic signal simulator according to claim 6, characterized in that, In step six, the frequency response of the compensation filter corresponding to each frame of the digital signal is calculated, and the digital signal is filtered as follows: (7.1) Calculate the magnitude of the gain of the compensation filter. : in, This represents the floor operation; (7.2) Calculate the phase of the filter gain : in, The set group delay constant, The imaginary unit, The number of points in the time-domain waveform; (7.3) The frequency response of the compensation filter is calculated. : (7.4) For signals Compensation filtering is performed to obtain the filtered signal. : in, , This represents the number of points in each frame of the time-domain waveform. The imaginary unit, To compensate for the frequency response of the filter.

8. The precise compensation method for an acoustic signal simulator according to claim 7, characterized in that, In step seven, each frame of the filtered digital signal is merged. For overlapping parts, the signals are merged by averaging. Non-overlapping parts are merged directly. The merged signal is used to drive the acoustic signal simulator to obtain an accurate acoustic signal.

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