Filter parameter determination method, apparatus, device, medium, and program product

Abstract

Embodiments of the present disclosure relate to a filter parameter determination method, apparatus, device, medium and program product. The method comprises: obtaining a plurality of impulse responses for each loudspeaker in an audio system, and determining a zero point parameter and a pole point parameter corresponding to the loudspeaker according to the plurality of impulse responses, wherein the plurality of impulse responses are determined according to received signals obtained by a plurality of pickup components receiving audio signals played by the loudspeaker; and determining filter coefficients corresponding to each target filter according to the zero point parameter and the pole point parameter corresponding to each loudspeaker, wherein the filter coefficients are used for the corresponding target filter to filter and process an input audio signal and output a low-frequency audio signal for the corresponding loudspeaker to play. The method can produce a real bass enhancement effect and avoid user auditory fatigue.

Description

Technical Field

[0001] This disclosure relates to the field of audio processing technology, and in particular to a method, apparatus, device, medium, and program product for determining filter parameters. Background Technology

[0002] Today, many users crave bass response when listening to music, and this applies to listening inside a vehicle as well. Therefore, bass enhancement technology is becoming increasingly important.

[0003] In existing bass enhancement technologies, the perceived bass is generated on the software side based on high-frequency harmonics from psychoacoustics. This technology produces high-frequency sound signals, but the user perceives them as low-frequency signals.

[0004] However, since the low-frequency sound perception produced by existing technology is generated psychologically by the user and has a virtual feel, the low-frequency sound signals produced lack realism and can lead to auditory fatigue. Summary of the Invention

[0005] This disclosure provides a method, apparatus, device, medium, and program product for determining filter parameters, which can be used to generate realistic bass enhancement effects and avoid user auditory fatigue.

[0006] In a first aspect, embodiments of this disclosure provide a filter parameter determination method for an audio system, the audio system including a speaker array and target filters corresponding to the speakers in the speaker array; the method includes:

[0007] For each speaker, multiple impulse responses are acquired, and the zero-point parameters and pole parameters corresponding to the speaker are determined based on these multiple impulse responses. The multiple impulse responses are determined based on the received signals obtained by multiple pickup components receiving the audio signals played by the speaker.

[0008] The filter coefficients corresponding to each target filter are determined based on the zero-point parameter and pole parameter corresponding to each speaker. These filter coefficients are used by the corresponding target filter to filter the input audio signal and output a low-frequency audio signal for playback by the corresponding speaker.

[0009] In one embodiment, obtaining multiple impulse responses includes: obtaining multiple initial impulse responses; and performing low-pass filtering on the multiple initial impulse responses to obtain the multiple impulse responses.

[0010] In one embodiment, acquiring multiple initial impulse responses includes: acquiring multiple received signals obtained by the multiple pickup components respectively receiving audio signals played by the speaker; the audio signal played by the speaker is a swept frequency signal, and each pickup component is arranged in the theoretical three-dimensional listening area corresponding to the speaker array; for each received signal, the received signal is convolved with the inverse swept frequency signal to obtain the initial impulse response corresponding to the received signal, wherein the swept frequency signal and the inverse swept frequency signal are inverses of each other.

[0011] In one embodiment, determining the zero-point parameters and pole parameters corresponding to the loudspeaker based on the plurality of impact responses includes: clustering the plurality of impact responses to obtain a target impact response; making an all-pole assumption on the target impact response to obtain an all-pole expression; solving the all-pole expression using a preset solving algorithm to obtain the coefficients of the all-pole expression, and using the coefficients of the all-pole expression as the pole parameters; and determining the zero-point parameters based on the pole parameters.

[0012] In one embodiment, determining the zero-point parameter based on the pole parameter includes: for each impact response, determining the Z-domain pole expression corresponding to the impact response based on the pole parameter, converting the Z-domain pole expression into a time-domain pole expression, convolving the time-domain pole expression with the impact response to obtain a time-domain zero-point expression; for each time-domain zero-point expression, performing a fitting process using a preset linear fitting method to obtain the zero-point parameter corresponding to the time-domain zero-point expression.

[0013] In one embodiment, determining the filter coefficients corresponding to each target filter based on the zero-point parameters and pole parameters corresponding to each loudspeaker includes: determining a first coefficient matrix based on the zero-point parameters corresponding to each loudspeaker; determining a second coefficient matrix based on the pole parameters corresponding to each loudspeaker; multiplying the first coefficient matrix and the second coefficient matrix to obtain a target coefficient matrix; and determining the filter coefficients corresponding to each target filter based on the elements included in the target coefficient matrix.

[0014] In one embodiment, the audio system further includes a high-pass filter and a low-pass filter. The method further includes: performing high-pass filtering on the original audio signal using the high-pass filter to obtain an initial first audio signal, and performing delay processing on the initial first audio signal to obtain a first audio signal; performing low-pass filtering on the original audio signal using the low-pass filter to obtain an initial second audio signal, and performing filtering processing on each initial second audio signal using each target filter to obtain a second audio signal output by each target filter; superimposing each second audio signal with the first audio signal to obtain multiple target audio signals, and outputting each target audio signal corresponding to each speaker of each target filter.

[0015] Secondly, embodiments of this disclosure provide a filter parameter determination apparatus for use in an audio system, the audio system including a speaker array and target filters corresponding to the speakers in the speaker array; the apparatus includes:

[0016] The first determining module is used to acquire multiple impulse responses for each of the loudspeakers, and determine the zero-point parameters and pole parameters corresponding to the loudspeakers based on the multiple impulse responses, wherein the multiple impulse responses are determined based on the received signals obtained by multiple pickup components receiving the audio signals played by the loudspeakers;

[0017] The second determining module is used to determine the filter coefficients corresponding to each of the target filters based on the zero-point parameters and pole parameters corresponding to each of the speakers respectively. The filter coefficients are used to filter the input audio signal and output a low-frequency audio signal for playback by the corresponding speaker.

[0018] In one embodiment, the first determining module is specifically used to: acquire multiple initial impulse responses; and perform low-pass filtering on the multiple initial impulse responses to obtain the multiple impulse responses.

[0019] In one embodiment, the first determining module is specifically used to: acquire multiple received signals obtained by the multiple pickup components respectively receiving the audio signal played by the loudspeaker; the audio signal played by the loudspeaker is a sweep frequency signal, and each pickup component is arranged in the theoretical three-dimensional listening area corresponding to the loudspeaker array; for each received signal, the received signal is convolved with the inverse sweep frequency signal to obtain the initial impulse response corresponding to the received signal, wherein the sweep frequency signal and the inverse sweep frequency signal are inverses of each other.

[0020] In one embodiment, the first determining module is specifically used for: clustering the multiple impact responses to obtain a target impact response; making an all-pole assumption on the target impact response to obtain an all-pole expression; using a preset solving algorithm to solve the all-pole expression to obtain the coefficients of the all-pole expression, and using the coefficients of the all-pole expression as the pole parameters; and determining the zero-point parameters based on the pole parameters.

[0021] In one embodiment, the first determining module is specifically used to: for each impact response, determine the Z-domain pole expression corresponding to the impact response based on the pole parameter, convert the Z-domain pole expression into a time-domain pole expression, and convolve the time-domain pole expression with the impact response to obtain a time-domain zero expression; for each time-domain zero expression, perform fitting processing using a preset linear fitting method to obtain the zero parameter corresponding to the time-domain zero expression.

[0022] In one embodiment, the second determining module is specifically used to: determine a first coefficient matrix based on the zero-point parameters corresponding to each loudspeaker; determine a second coefficient matrix based on the pole parameters corresponding to each loudspeaker; multiply the first coefficient matrix and the second coefficient matrix to obtain a target coefficient matrix; and determine the filter coefficients corresponding to each target filter based on the elements included in the target coefficient matrix.

[0023] In one embodiment, the audio system further includes a high-pass filter and a low-pass filter, and the device also includes:

[0024] The first filtering module is used to perform high-pass filtering on the original audio signal to obtain an initial first audio signal, and to perform delay processing on the initial first audio signal to obtain a first audio signal.

[0025] The second filtering module is used to perform low-pass filtering on the original audio signal to obtain an initial second audio signal, and to use each target filter to filter each initial second audio signal to obtain the second audio signal output by each target filter.

[0026] The output module is used to superimpose each second audio signal with the first audio signal to obtain multiple target audio signals, and to output each target audio signal from each speaker corresponding to each target filter.

[0027] Thirdly, embodiments of this disclosure provide a computer device including a memory, a processor, and a computer program stored in the memory and executable on the processor, wherein the processor executes the computer program to implement the method described in the first aspect above.

[0028] Fourthly, embodiments of this disclosure provide a computer-readable storage medium having a computer program stored thereon, which, when executed by a processor, implements the method described in the first aspect above.

[0029] Fifthly, embodiments of this disclosure provide a computer program product, including a computer program that, when executed by a processor, implements the method described in the first aspect above.

[0030] The filter parameter determination method, apparatus, device, medium, and program product provided in this disclosure can be used in an audio system including a speaker array and target filters corresponding to the speakers in the speaker array. Specifically, for each speaker, multiple impulse responses can be acquired, and the zero-point parameters and pole parameters corresponding to the speaker can be determined based on these multiple impulse responses. These multiple impulse responses are determined based on received signals obtained from the audio signals played by the speakers by multiple pickup components. Further, the filter coefficients corresponding to each target filter can be determined based on the zero-point parameters and pole parameters corresponding to each speaker. These filter coefficients can then be used by the corresponding target filters to filter the input audio signals and output low-frequency audio signals for playback by the corresponding speakers, thereby achieving bass enhancement. Specifically, by accurately determining the filter coefficients of each target filter through the zero-point parameters and pole parameters of each speaker, the audio system can achieve different degrees of bass enhancement during application by filtering the input audio signals through each target filter. This allows the speakers to collaboratively play bass audio signals, increasing the bass enhancement capability of the audio system. Because the bass audio signal is generated by filtering each speaker filter and then played through each speaker, the bass produced by this audio system is realistic. While increasing the bass feel, it does not have a virtual feel and will not cause auditory fatigue for the user. Attached Figure Description

[0031] Figure 1 This is a flowchart illustrating a filter parameter determination method in one embodiment;

[0032] Figure 2 This is a flowchart illustrating the process of obtaining multiple impact responses in one embodiment;

[0033] Figure 3 This is a flowchart illustrating the process of obtaining multiple initial impact responses in one embodiment;

[0034] Figure 4 This is a flowchart illustrating the process of determining zero-point and pole parameters in one embodiment;

[0035] Figure 5 This is a flowchart illustrating the process of determining zero-point parameters in one embodiment;

[0036] Figure 6 This is a flowchart illustrating the process of determining the filter coefficients corresponding to each target filter in one embodiment.

[0037] Figure 7 This is a schematic diagram illustrating the process of processing and playing the original audio signal in one embodiment;

[0038] Figure 8 This is a logical diagram illustrating the processing procedure of a frequency conversion system in one embodiment;

[0039] Figure 9 This is a flowchart illustrating the process of determining loudspeaker filter coefficients in one embodiment;

[0040] Figure 10 This is a structural block diagram of a filter parameter determination device in one embodiment;

[0041] Figure 11 This is an internal structural diagram of a computer device in one embodiment. Detailed Implementation

[0042] To make the objectives, technical solutions, and advantages of the embodiments of this disclosure clearer, the embodiments of this disclosure will be further described in detail below with reference to the accompanying drawings and examples. It should be understood that the specific embodiments described herein are merely illustrative of the embodiments of this disclosure and are not intended to limit the embodiments of this disclosure.

[0043] First, before introducing the technical solutions of the embodiments of this disclosure in detail, let's first introduce the technical background or evolution of the embodiments of this disclosure. Generally, in the field of bass enhancement technology, the current technical background is as follows: a powerful bass response is what most users seek when listening to music. The stronger the bass, the larger the size of the speaker playing the audio signal, and the higher the speaker's power consumption. Currently, in the field of electric vehicles, electric vehicles have strict limitations on interior space and power consumption; only by using ordinary small-sized speakers can excessive power consumption be avoided. However, this results in a significant attenuation of bass in the audio signal played by the speaker. Therefore, bass enhancement technology is needed. Currently, bass enhancement is usually achieved through software, specifically by using high-frequency harmonics in psychoacoustics to generate a bass response. Because this technology generates high-frequency sound signals, but users psychologically perceive them as low-frequency sound signals. Based on this background, through long-term research and the collection, demonstration, and verification of experimental data, the applicant has found that the bass audio signals generated by existing technologies have a virtual feel; correspondingly, the low-frequency sound signals they generate lack realism and can lead to auditory fatigue in users. Therefore, providing a bass enhancement technology that generates realistic bass audio signals has become a pressing problem to be solved. Furthermore, it should be noted that the applicant has devoted considerable creative effort to identifying the lack of realism in audio signals generated by existing technologies and the resulting auditory fatigue, as well as to the technical solutions described in the following embodiments.

[0044] The technical solutions involved in the embodiments of this disclosure will be described below in conjunction with the scenarios in which they are applied.

[0045] In one embodiment, such as Figure 1 As shown, a method for determining filter parameters is provided, with an example of its application in an audio system. The audio system includes at least a speaker array consisting of multiple speakers and a target filter corresponding to each speaker. Optionally, the audio system may include a computer device to determine the filter parameters of each target filter and control the target filters to filter the audio signal before playback through the corresponding speakers. The computer device may be, but is not limited to, various personal computers, laptops, smartphones, tablets, audio playback devices, and servers. Furthermore, the number of speakers and target filters is not specifically limited in this embodiment. Optionally, the audio system may be located in the center console of a vehicle or other locations, or it may be located in other devices; this embodiment does not specifically limit this. The filter parameter determination method includes the following steps:

[0046] Step 101: For each speaker, acquire multiple impulse responses, and determine the zero-point parameters and pole parameters corresponding to the speaker based on the multiple impulse responses. The multiple impulse responses are determined based on the received signals obtained by multiple pickup components receiving the audio signals played by the speaker.

[0047] To enable the audio system to output realistic bass audio signals and achieve bass enhancement, the audio system may include multiple speakers and target filters corresponding to each speaker. The original audio signal is filtered by each target filter and then output through its corresponding speaker, allowing the user to hear the bass enhancement achieved collaboratively by the speakers. To ensure that each target filter accurately filters the original signal to generate realistic low-frequency audio signals and achieve bass enhancement, the filter coefficients of each target filter need to be predetermined. Optionally, in this embodiment, the target filter can be a speaker filter, which can be used to improve the pitch and frequency of the audio signal, and the filter coefficients of the speaker filter determine the characteristics of the improved audio. Therefore, it is necessary to obtain the filter coefficients of each target filter.

[0048] For each loudspeaker, the filter coefficients of the target filter corresponding to that loudspeaker can be determined based on multiple impulse responses of that loudspeaker. The impulse response characterizes the acoustic transmission path of the loudspeaker to the preset spatial region, and can be used to characterize the frequency and phase changes of the audio signal output by the loudspeaker after reaching the preset spatial region. By acquiring multiple different impulse responses of the loudspeaker, the acoustic transmission path of the loudspeaker to different points in the preset spatial region is determined. Based on these multiple impulse responses, the coefficients of the target filter corresponding to that loudspeaker are further determined, so that the audio signal filtered by the target filter achieves a stereo bass effect after propagating to the preset spatial region.

[0049] Specifically, before determining the filter coefficients of the target filter, multiple pickup components can be arranged in a preset spatial area. A preset audio signal is played using the speaker, and multiple impulse responses corresponding to the speaker are determined based on the received signals received by each pickup component. In this embodiment, the number of pickup components is not specifically limited, and the number of pickup components corresponds to the number of impulse responses obtained. Optionally, the pickup components can be microphones or other audio receiving devices.

[0050] For each loudspeaker, since the zeros and poles of the impulse response are important parameters for determining the filter coefficients, and the zero parameters refer to the coefficients of the zero-point expression and the pole parameters are the coefficients of the pole expression, multiple impulse responses corresponding to the loudspeaker can be obtained, and the zero parameters and pole parameters corresponding to the loudspeaker can be determined based on these multiple impulse responses. In other words, the zero parameters and pole parameters also correspond to the target filter corresponding to the loudspeaker.

[0051] Therefore, the zero-point parameters and pole parameters corresponding to each loudspeaker can be obtained, and the filter coefficients of the target filter corresponding to each loudspeaker can be determined based on these zero-point parameters and pole parameters.

[0052] Step 102: Determine the filter coefficients corresponding to each target filter based on the zero-point parameter and pole parameter corresponding to each speaker. These filter coefficients are used by the corresponding target filter to filter the input audio signal and output a low-frequency audio signal for playback by the corresponding speaker.

[0053] Specifically, for each loudspeaker, the zero-point expression of its impulse response can be determined based on the zero-point parameters corresponding to that loudspeaker, and the pole expression of its impulse response can be determined based on the pole parameters corresponding to that loudspeaker. Optionally, a zero-point coefficient matrix can be determined based on the zero-point expressions corresponding to each loudspeaker, and a pole coefficient matrix can be determined based on the pole expressions corresponding to each loudspeaker. A coefficient matrix containing the filter coefficients corresponding to each target filter can then be obtained based on the elements in the coefficient matrix.

[0054] Each filter coefficient is applied to the corresponding target filter so that when the audio system needs to output an audio signal, the audio signal is filtered by each target filter and output through the corresponding speaker, so that the audio played by each speaker is a real signal containing low-frequency audio, with a bass impact and achieving a bass enhancement effect.

[0055] The aforementioned filter parameter determination method can be used in audio systems comprising a speaker array and target filters corresponding to the speakers in the speaker array. Specifically, for each speaker, multiple impulse responses can be acquired, and the zero-point parameters and pole parameters corresponding to that speaker can be determined based on these multiple impulse responses. These multiple impulse responses are determined based on received signals obtained from the audio signals played by the speakers by multiple pickup components. Further, the filter coefficients corresponding to each target filter can be determined based on the zero-point parameters and pole parameters corresponding to each speaker. These filter coefficients can then be used by the corresponding target filters to filter the input audio signal and output a low-frequency audio signal for playback by the corresponding speakers, thus achieving bass enhancement. Specifically, by accurately determining the filter coefficients of each target filter through the zero-point parameters and pole parameters of each speaker, the audio system can achieve different degrees of bass enhancement during application by filtering the input audio signal through each target filter. Furthermore, the speakers can collaboratively play bass audio signals, increasing the bass enhancement capability of the audio system. Because the bass audio signal is generated by filtering each speaker filter and then played through each speaker, the bass produced by this audio system is realistic. While increasing the bass feel, it does not have a virtual feel and will not cause auditory fatigue for the user.

[0056] As mentioned above, multiple impulse responses need to be obtained for each loudspeaker in order to determine the zero-point and pole parameters. The process of obtaining multiple impulse responses for each loudspeaker will be explained below.

[0057] In one embodiment, such as Figure 2 The diagram illustrates a process for obtaining multiple impact responses according to an embodiment of this application. For each speaker, obtaining multiple impact responses includes:

[0058] Step 201: Obtain multiple initial impact responses.

[0059] Multiple initial impulse responses can be determined by receiving multiple received signals from the audio signals played by the speaker by multiple pickup components. These initial impulse responses may be affected by other interference factors. Therefore, these multiple initial impulse responses are preliminarily processed to obtain the multiple impulse responses.

[0060] Please refer to Figure 3 This illustration shows a flowchart of obtaining multiple initial impact responses according to an embodiment of this application. For each speaker, obtaining multiple initial impact responses includes:

[0061] Step 301: Obtain multiple received signals obtained by the multiple pickup components respectively receiving the audio signal played by the speaker; the audio signal played by the speaker is a frequency sweep signal, and each pickup component is arranged in the theoretical three-dimensional listening area corresponding to the speaker array.

[0062] Taking an example where the multiple sound pickup components consist of n microphones, n is greater than or equal to 4, and the audio system is located in the central control position of an electric vehicle, the n microphones are arranged within a theoretical three-dimensional listening area, forming a stereo microphone array. This theoretical three-dimensional listening area refers to the user's hearing range when inside the vehicle, such as the area around the ear. This theoretical three-dimensional listening area is the preset spatial area mentioned above.

[0063] Each speaker plays the sweep frequency signal as an audio signal, and correspondingly, n microphones receive the audio signal played by each speaker to obtain n received signals. It should be noted that this application embodiment does not specifically limit the number or type of pickup devices.

[0064] Step 302: For each received signal, the received signal is convolved with the inverse sweep frequency signal to obtain the initial impulse response corresponding to the received signal. The sweep frequency signal and the inverse sweep frequency signal are inverses of each other.

[0065] The multiple initial impulse responses can be obtained by convolving each received signal with the inverse frequency sweep signal. Frequency sweep refers to the continuous change of signal frequency from high to low (or from low to high) within a frequency band. The inverse frequency sweep signal is the inverse of the frequency sweep signal.

[0066] For the i-th speaker in the speaker array, the received signal obtained by the j-th microphone receiving the audio signal played by the i-th speaker is denoted as s(i,j), and the inverse frequency sweep signal is denoted as inv_sweep. The received signal and the inverse frequency sweep signal are convolved to obtain the initial impulse response corresponding to the received signal, that is, the initial impulse response of the i-th speaker to the j-th microphone. This represents convolution. Therefore, for the i-th speaker, its corresponding n initial impulse responses can be obtained.

[0067] Step 202: Perform low-pass filtering on the multiple initial impulse responses to obtain the multiple impulse responses.

[0068] In this embodiment, the main principle of bass enhancement is to utilize the cooperation of multiple speakers. By using the bass wavelength, the driver's auditory area (the theoretical three-dimensional listening area) is controlled to ensure that the phases of the arriving low-frequency sound signals are approximately consistent, thus achieving bass enhancement. However, on the one hand, audio signals have a spatial sense but lack directionality within a certain frequency range; on the other hand, higher frequency audio signals have poor control and are prone to confusion, both affecting the realization of bass enhancement. Therefore, in this embodiment, to determine more accurate filter coefficients, low-pass filtering is required for each initial impulse response to obtain an impulse response that conforms to bass enhancement theory.

[0069] Optionally, for each speaker, a low-pass filter can be used to perform low-pass filtering on each initial impulse response. Optionally, based on considerations of bass directionality and control, the cutoff frequency of the low-pass filter can be set between 300 Hz and 500 Hz; this embodiment of the application does not specifically limit this. Thus, n impulse responses corresponding to each speaker can be obtained.

[0070] In this embodiment, for each speaker, multiple initial impulse responses are determined based on multiple received signals obtained from the audio signals played by the speaker by multiple pickup components. These initial impulse responses are then low-pass filtered to obtain the multiple impulse responses. Target filter coefficients are determined based on these low-pass filtered impulse responses, which improves the accuracy of the determined filter coefficients. This allows the target filter to process audio signals more accurately during application, thereby enhancing the bass enhancement effect of the audio system. Furthermore, since each pickup component is arranged within the theoretical three-dimensional listening area corresponding to the speaker array, it can fully simulate the audio signals audible to the user. This ensures that after determining the filter coefficients of the target filter, the audio output by each speaker in the audio system better matches the user's hearing range.

[0071] As mentioned above, for a single loudspeaker, its corresponding zero-point and pole parameters can be determined based on multiple impulse responses. The process of determining multiple zero-point and pole parameters will be explained below.

[0072] In one embodiment, such as Figure 4 The diagram illustrates a flowchart of a method for determining zero-point and pole parameters according to an embodiment of this application. Based on the multiple impulse responses, the zero-point and pole parameters corresponding to the loudspeaker are determined, including:

[0073] Step 401: Cluster the multiple impact responses to obtain the target impact response.

[0074] Here, h(i,j) in the above text is the time-domain expression of the impulse response. However, solving for the filter coefficients of each filter has stronger control in the Z-domain; therefore, the time-domain expression can be transformed into a Z-domain expression. Generally, the impulse response can be expressed in either the time domain or the Z-domain. Assume the Z-domain expression H(z) of the impulse response is:

[0075]

[0076] In the above formula, A(z) is the zero-point expression of the impulse response, and B(z) is the pole-point expression of the impulse response. The zero-point expression can be determined by determining the zero-point parameters, and the pole-point expression can be determined by determining the pole-point parameters. Furthermore, as mentioned above, the filter coefficient matrix can be determined based on the zero-point and pole-point expressions corresponding to each loudspeaker, thus obtaining the filter coefficients of each target filter.

[0077] For each loudspeaker, the impulse response corresponds to n microphones. Here, poles represent the commonalities of the impulse responses, and zeros represent the differences. In other words, the impulse responses share a common pole but have different zeros. Based on this, for each loudspeaker, the pole expression can be determined through clustering and fitting. Due to the common poles, the pole expression for the impulse response of each microphone is the same, and this pole expression can be obtained by determining the pole parameters.

[0078] Specifically, for each loudspeaker, the target impulse response for that loudspeaker is obtained by clustering multiple impulse responses. The pole parameters are then further determined based on this impulse response.

[0079] Step 402: Apply the all-pole assumption to the target impact response to obtain the all-pole expression.

[0080] Specifically, to obtain the pole parameters, a full pole assumption can be made on the target impact response obtained after clustering. Optionally, the LPC (Linear Predictive Coding) fitting method can be used to find common poles to determine the pole parameters. The LPC fitting method assumes that the value of a discrete function h(n) can be linearly expressed by the previous discrete value h(ni). Therefore, the full pole expression obtained by applying the LPC fitting method to the target impact response with a full pole assumption is as follows:

[0081]

[0082] In the above formula, h(n) is the target impact response; k is the order of the poles; b i are the coefficients of the all-pole expression, i.e., the pole parameters; i is the intermediate parameter.

[0083] Given the target impact response, the pole parameters can be obtained by solving the above all-pole expression.

[0084] Optionally, other fitting methods can be used to make the full pole assumption on the target impact response. In this embodiment, the LPC fitting method is used as an example, but it is not limited to this method.

[0085] Step 403: The all-pole expression is solved using a preset solution algorithm to obtain the coefficients of the all-pole expression, and the coefficients of the all-pole expression are used as the pole parameters.

[0086] Optionally, the Burg algorithm can be used as the preset solution algorithm to solve the above all-pole expression and obtain the pole parameters b. i The Burger algorithm is a recursive algorithm that directly calculates the power spectrum estimate from a known time signal sequence. Optionally, in this embodiment, other algorithms can also be used to solve the above all-pole expression, and this embodiment does not specifically limit the preset solution algorithm.

[0087] Step 404: Determine the zero-point parameter based on the pole parameter.

[0088] Specifically, after determining the pole parameters, the pole expression B(z) can be determined. Given the pole expression and the impulse response, the zero parameters can then be determined to obtain the zero expressions for each impulse response.

[0089] Please refer to Figure 5 The diagram illustrates a flowchart of a method for determining zero-point parameters according to an embodiment of this application. Determining the zero-point parameters based on the pole parameters includes:

[0090] Step 501: For each impulse response, determine the Z-domain pole expression corresponding to the impulse response based on the pole parameter.

[0091] For each impulse response, since there is a common pole, once the pole parameters are determined, the pole expression corresponding to each impulse response can be obtained, that is, the Z-domain pole expression B(z).

[0092] Step 502: Convert the Z-domain pole expression into a time-domain pole expression, and convolve the time-domain pole expression with the impulse response to obtain a time-domain zero expression.

[0093] The Z-domain pole expression is used as a zero and converted to a time-domain pole expression h_b(n). Since the impulse responses obtained above are also time-domain expressions, for each impulse response, convolving the impulse response with the time-domain pole expression h_b(n) yields its corresponding time-domain zero expression h_a(n). Solving the time-domain zero expression yields the zero-point parameters of each impulse response, thus determining the zero-point expression.

[0094] Step 503: For each time-domain zero-point expression, a preset linear fitting method is used for fitting to obtain the zero-point parameter corresponding to the time-domain zero-point expression.

[0095] As mentioned above, each loudspeaker has different impulse response zeros. Given that the time-domain expression for each zero is determined to be the impulse response convolution time-domain pole expression, a pre-defined linear fitting method can be used for fitting to obtain the zero-point parameter 'a' corresponding to that time-domain zero-point expression. i Zero-point parameter a i These are the coefficients of the zero-point expression A(z) above. Therefore, for each impact response, based on a i The zero-point expression A(z) can be determined. Thus, the transformation from the time-domain impulse response to the z-domain impulse response H(z) is also completed.

[0096] Optionally, the preset linear fitting method can be the moving average model (MA) linear fitting method. Of course, it can also be other linear fitting methods. This application embodiment does not specifically limit this.

[0097] In this embodiment, for each loudspeaker, based on the common poles and different zeros of each loudspeaker's impulse response, the target impulse response is obtained by clustering multiple impulse responses. Based on the target impulse response, the pole parameters can be obtained by making an all-pole assumption. Based on the pole expression determined by the pole parameters, the zero parameters are further determined to determine the zero expression, thus providing a basis for determining more accurate and reasonable filter coefficients.

[0098] For an audio system, it is known that through the above process, one pole parameter and n zero parameters corresponding to each speaker can be determined, thus determining one pole expression and n zero expressions, and therefore determining the filter coefficients. The process of determining the filter coefficients corresponding to each target filter will be explained below.

[0099] In one embodiment, such as Figure 6The diagram illustrates a flowchart of an embodiment of this application for determining the filter coefficients corresponding to each target filter. Determining the filter coefficients corresponding to each target filter based on the zero-point parameter and pole parameter corresponding to each loudspeaker includes:

[0100] Step 601: Determine the first coefficient matrix based on the zero-point parameters corresponding to each loudspeaker.

[0101] Step 602: Determine the second coefficient matrix based on the pole parameters corresponding to each loudspeaker.

[0102] Step 603: Multiply the first coefficient matrix by the second coefficient matrix to obtain the target coefficient matrix; determine the filter coefficients corresponding to each target filter based on the elements included in the target coefficient matrix.

[0103] As mentioned above, for each loudspeaker, n zero-point expressions can be determined based on n zero-point parameters. Let the audio system contain M loudspeakers; then we can obtain n*M zero-point expressions A. 11 (z)…A nM (z). For each loudspeaker, a pole expression can be obtained; for M loudspeakers, M pole expressions B1(z)…B can be obtained. M (z).

[0104] The ultimate goal of low-frequency enhancement is to ensure that the frequency of the audio signals output by each speaker is consistent within the theoretical three-dimensional listening range, i.e., the user's hearing range, and that after some filtering, the audio signals played by each speaker are identical to the original audio signals without attenuation. In this embodiment, the least-means method can be used to solve for the filter coefficients; specifically, the least squares method or other methods within the least-means method can be used to solve for the filter coefficients.

[0105] Let H be the impact response of the i-th speaker to the theoretical three-dimensional listening area. i (z), let G be the filter coefficient of the target filter corresponding to the i-th loudspeaker. i (z), then the following constraints can be obtained:

[0106]

[0107] Where argminf(x) represents the value of the variable that minimizes the objective function f(x); M is the total number of loudspeakers; and * indicates multiplication.

[0108] Based on the above constraints, we can obtain: G(z) = B*A -1 ,in:

[0109]

[0110]

[0111] In the above formula, A -1 B is the first coefficient matrix; B is the second coefficient matrix. i (z) is the pole expression for the impulse response of the i-th loudspeaker, i = 1, 2, ..., M. A nM (z) is the zero-point expression for the impulse response of the Mth loudspeaker to the nth pickup component. n is the total number of pickup components; M is the total number of loudspeakers, which is also the total number of target filters.

[0112] Furthermore, numerical methods, analytical methods, and other solution methods are used to solve B*A. -1 This yields the target coefficient matrix, whose elements are the filter coefficients corresponding to each target filter, thus determining the filter coefficients of each target filter.

[0113] Furthermore, in this embodiment, the principle of low-frequency enhancement is that the wave emitted by the speaker is the enhancement region in the theoretical three-dimensional listening area. The audio signal received by the user in this theoretical three-dimensional listening area is the sum of the audio played by all the speakers. Therefore, the audio signal Y(z) received in this theoretical three-dimensional listening area is: X i (z) represents the original audio signal played by the i-th speaker.

[0114] In this embodiment, the hardware characteristics and long-wavelength characteristics of the audio system are fully utilized. The filter coefficients of each target filter are obtained by solving the zero-point expression and pole expression of the impulse response of each speaker. The filter coefficients of each target filter are then controlled to control a theoretical three-dimensional listening area as a bass enhancement area. Taking the audio system located in an electric vehicle as an example, this theoretical three-dimensional listening area can be the area where the user's head is located in the driving position. Thus, bass enhancement is achieved based on real low-frequency audio signals, avoiding auditory fatigue.

[0115] Typically, the audio played by an audio system includes not only low-frequency audio but also high-frequency audio. Therefore, the audio system also includes high-pass and low-pass filters to process audio signals of different frequencies and achieve excellent bass enhancement.

[0116] In one embodiment, such as Figure 7 The diagram illustrates a flowchart of raw audio signal processing and playback provided in an embodiment of this application. The audio system also includes a high-pass filter and a low-pass filter, and the method for determining the filter parameters further includes:

[0117] Step 701: Use the high-pass filter to perform high-pass filtering on the original audio signal to obtain an initial first audio signal, and perform delay processing on the initial first audio signal to obtain a first audio signal.

[0118] To achieve a pure bass enhancement effect, allowing the target filter to filter only the low-frequency original audio signal, it's necessary to distinguish between high-frequency and low-frequency audio in the original audio signal. Optionally, based on the characteristics of low-frequency audio, the cutoff frequency of the high-pass filter can be set to 300Hz. This high-pass filter is used to perform high-pass filtering on the original audio signal to obtain the initial first audio signal. Since the low-frequency audio requires a certain amount of time for low-pass filtering and filtering by each target filter, a delay is needed to ensure synchronized playback of low-frequency and high-frequency audio, resulting in the first audio signal output simultaneously with the target filter.

[0119] Step 702: Use the low-pass filter to perform low-pass filtering on the original audio signal to obtain the initial second audio signal, and use each target filter to filter each initial second audio signal to obtain the second audio signal output by each target filter.

[0120] As mentioned above, to achieve a pure bass enhancement effect and allow the target filter to filter only the low-frequency original audio signal, it is necessary to distinguish between high-frequency and low-frequency audio in the original audio signal. Therefore, the original audio signal is low-pass filtered to obtain an initial second audio signal containing low-frequency audio, and each initial second audio signal is filtered separately using each target filter. Optionally, the cutoff frequency of the low-pass filter can be 300Hz.

[0121] Step 703: Each second audio signal is superimposed on the first audio signal to obtain multiple target audio signals, and each target audio signal is output by the corresponding speaker of each target filter.

[0122] The second audio signal output from each target filter is then superimposed on the first audio signal to obtain the final audio signal, which can be directly used by the speaker corresponding to that target filter for playback. The speakers work together to enhance bass and achieve a realistic bass boost effect. Please refer to [reference needed]. Figure 8This diagram illustrates a logical schematic of an audio system processing procedure provided in an embodiment of this application. Taking the target filters as examples of speaker filters, the original audio is filtered by a low-pass filter and a high-pass filter, both with a cutoff frequency of 300Hz. The low-frequency audio processed by the low-pass filter is then filtered by the speaker filter to obtain low-frequency audio, while the high-pass filter outputs high-frequency audio. To achieve audio synchronization, the high-frequency audio is delayed and then superimposed with the low-frequency audio to obtain the final audio, which is then output by the speaker. Optionally, the delay processing can be implemented using a delay component; this embodiment of the application does not specifically limit the delay component.

[0123] In one embodiment, such as Figure 9 The diagram illustrates a flowchart of a method for determining loudspeaker filter coefficients according to an embodiment of this application. The process includes:

[0124] Step 901: Obtain multiple received signals obtained by receiving audio signals played by the speaker from multiple microphones.

[0125] The audio signal played by the speaker is a frequency sweep signal, and each microphone is arranged in the theoretical three-dimensional listening area corresponding to the speaker array.

[0126] Step 902: For each received signal, the received signal is convolved with the inverse frequency sweep signal to obtain the initial impulse response corresponding to the received signal.

[0127] Among them, the sweep frequency signal and the inverse sweep frequency signal are inverses of each other.

[0128] Step 903: Perform low-pass filtering on multiple initial impulse responses to obtain multiple impulse responses.

[0129] The cutoff frequency of the low-pass filter used for low-pass filtering is 300 Hz.

[0130] Step 904: Cluster the multiple impact responses to obtain the target impact response.

[0131] Step 905: Assume all poles on the target impact response to obtain an all pole expression, and use a preset solution algorithm to solve the all pole expression to obtain the pole parameters.

[0132] The preset solution algorithm is the LPC fitting method.

[0133] Step 906: For each impulse response, determine the Z-domain pole expression corresponding to the impulse response based on the pole parameters, convert the Z-domain pole expression into a time-domain pole expression, and convolve the time-domain pole expression with the impulse response to obtain the time-domain zero expression.

[0134] Step 907: For each time-domain zero-point expression, a preset linear fitting method is used for fitting to obtain the zero-point parameters corresponding to the time-domain zero-point expression.

[0135] Among them, the preset linear fitting method is the linear fitting method of the ma model.

[0136] Step 908: Determine the zero-point coefficient matrix based on the zero-point parameters corresponding to each loudspeaker, and determine the pole-point coefficient matrix based on the pole parameters corresponding to each loudspeaker.

[0137] Step 909: Multiply the negative first power of the zero coefficient matrix with the pole coefficient matrix to obtain the target coefficient matrix. Determine the filter coefficients corresponding to each loudspeaker filter based on the elements included in the target coefficient matrix.

[0138] In this embodiment, based on the common pole and zero-point representation of multiple loudspeakers, the filter coefficients of each loudspeaker are calculated to enhance bass, thus determining the loudspeaker filter coefficients. By utilizing the cooperation of multiple loudspeakers and the bass wavelength, the audio signal is filtered based on the filters of each loudspeaker, ensuring that the bass phase reaching the user's head area in the driver's position is approximately consistent, achieving bass enhancement. The response of the played audio signal to the user's head area is consistent, and the received audio signal is exactly the same as the original audio source played, without attenuation.

[0139] It should be understood that although the steps in the flowcharts of the embodiments described above are shown sequentially according to the arrows, these steps are not necessarily executed in the order indicated by the arrows. Unless explicitly stated herein, there is no strict order restriction on the execution of these steps, and they can be executed in other orders. Moreover, at least some steps in the flowcharts of the embodiments described above may include multiple steps or multiple stages. These steps or stages are not necessarily completed at the same time, but can be executed at different times. The execution order of these steps or stages is not necessarily sequential, but can be performed alternately or in turn with other steps or at least some of the steps or stages of other steps.

[0140] In one embodiment, such as Figure 10 As shown, a filter parameter determination device 1000 is provided, comprising: a first determination module 1001 and a second determination module 1002, wherein:

[0141] The first determining module 1001 is used to acquire multiple impulse responses for each of the loudspeakers, and determine the zero-point parameters and pole parameters corresponding to the loudspeakers based on the multiple impulse responses, wherein the multiple impulse responses are determined based on the received signals obtained by multiple pickup components receiving the audio signals played by the loudspeakers.

[0142] The second determining module 1002 is used to determine the filter coefficients corresponding to each of the target filters according to the zero-point parameters and pole parameters corresponding to each of the speakers respectively. The filter coefficients are used to filter the input audio signal and output a low-frequency audio signal for playback by the corresponding speaker.

[0143] In one embodiment, the first determining module 1001 is specifically used to: acquire multiple initial impulse responses; and perform low-pass filtering on the multiple initial impulse responses to obtain the multiple impulse responses.

[0144] In one embodiment, the first determining module 1001 is specifically configured to: acquire multiple received signals obtained by the multiple pickup components respectively receiving the audio signal played by the loudspeaker; the audio signal played by the loudspeaker is a sweep frequency signal, and each pickup component is arranged in the theoretical three-dimensional listening area corresponding to the loudspeaker array; for each received signal, the received signal is convolved with the inverse sweep frequency signal to obtain the initial impulse response corresponding to the received signal, wherein the sweep frequency signal and the inverse sweep frequency signal are inverses of each other.

[0145] In one embodiment, the first determining module 1001 is specifically used for: clustering the multiple impact responses to obtain a target impact response; making an all-pole assumption on the target impact response to obtain an all-pole expression; solving the all-pole expression using a preset solving algorithm to obtain the coefficients of the all-pole expression, and using the coefficients of the all-pole expression as the pole parameters; and determining the zero-point parameters based on the pole parameters.

[0146] In one embodiment, the first determining module 1001 is specifically configured to: for each impact response, determine the Z-domain pole expression corresponding to the impact response based on the pole parameters, convert the Z-domain pole expression into a time-domain pole expression, convolve the time-domain pole expression with the impact response to obtain a time-domain zero expression; for each time-domain zero expression, perform fitting processing using a preset linear fitting method to obtain the zero parameters corresponding to the time-domain zero expression.

[0147] In one embodiment, the second determining module 1002 is specifically used to: determine a first coefficient matrix based on the zero-point parameters corresponding to each loudspeaker; determine a second coefficient matrix based on the pole parameters corresponding to each loudspeaker; multiply the first coefficient matrix and the second coefficient matrix to obtain a target coefficient matrix; and determine the filter coefficients corresponding to each target filter based on the elements included in the target coefficient matrix.

[0148] In one embodiment, the audio system further includes a high-pass filter and a low-pass filter, and the device further includes:

[0149] The first filtering module is used to perform high-pass filtering on the original audio signal to obtain an initial first audio signal, and to perform delay processing on the initial first audio signal to obtain a first audio signal.

[0150] The second filtering module is used to perform low-pass filtering on the original audio signal to obtain an initial second audio signal, and to use each target filter to filter each initial second audio signal to obtain the second audio signal output by each target filter.

[0151] The output module is used to superimpose each second audio signal with the first audio signal to obtain multiple target audio signals, and to output each target audio signal from each speaker corresponding to each target filter.

[0152] Specific limitations regarding the filter parameter determination device can be found in the limitations of the filter parameter determination method described above, and will not be repeated here. Each module in the aforementioned filter parameter determination device can be implemented entirely or partially through software, hardware, or a combination thereof. These modules can be embedded in or independent of the processor in a computer device in hardware form, or stored in the memory of an electronic device in software form, so that the processor can call and execute the operations corresponding to each module.

[0153] Figure 11 This is a block diagram illustrating a server 1400 according to an exemplary embodiment. (Refer to...) Figure 11 Server 1400 includes processing component 1420, which further includes one or more processors, and memory resources represented by memory 1422 for storing instructions or computer programs, such as application programs, that can be executed by processing component 1420. The application programs stored in memory 1422 may include one or more modules, each corresponding to a set of instructions. Furthermore, processing component 1420 is configured to execute instructions to perform the aforementioned filter parameter determination method.

[0154] Server 1400 may also include a power supply component 1424 configured to perform power management of device 1400, a wired or wireless network interface 1426 configured to connect device 1400 to a network, and an input / output (I / O) interface 1428. Server 1400 can operate on an operating system stored in memory 1422, such as Windows 14 Server™, Mac OS X™, Unix™, Linux™, FreeB14D™, or similar.

[0155] In an exemplary embodiment, a storage medium including instructions is also provided, such as a memory 1422 including instructions, which can be executed by the processor of server 1400 to perform the above-described method. The storage medium may be a non-transitory computer-readable storage medium, such as a ROM, random access memory (RAM), CD-ROM, magnetic tape, floppy disk, and optical data storage device.

[0156] In an exemplary embodiment, a computer program product is also provided, which, when executed by a processor, can implement the above-described methods. The computer program product includes one or more computer instructions. When these computer instructions are loaded and executed on a computer, some or all of the above-described methods can be implemented, wholly or partially, according to the processes or functions described in the embodiments of this disclosure.

[0157] Those skilled in the art will understand that all or part of the processes in the methods of the above embodiments can be implemented by a computer program instructing related hardware. The computer program can be stored in a non-volatile computer-readable storage medium, and when executed, it can include the processes of the embodiments of the above methods. Any references to memory, storage, databases, or other media used in the embodiments provided in this disclosure can include at least one of non-volatile and volatile memory. Non-volatile memory can include read-only memory (ROM), magnetic tape, floppy disk, flash memory, or optical storage, etc. Volatile memory can include random access memory (RAM) or external cache memory. By way of illustration and not limitation, RAM can be in various forms, such as static random access memory (SRAM) or dynamic random access memory (DRAM), etc.

[0158] The technical features of the above embodiments can be combined in any way. For the sake of brevity, not all possible combinations of the technical features in the above embodiments are described. However, as long as there is no contradiction in the combination of these technical features, they should be considered to be within the scope of this specification.

[0159] The above-described embodiments are merely illustrative of several implementation methods of the present disclosure, and while the descriptions are specific and detailed, they should not be construed as limiting the scope of the invention patent. It should be noted that those skilled in the art can make various modifications and improvements without departing from the concept of the present disclosure embodiments, and these all fall within the protection scope of the present disclosure embodiments. Therefore, the protection scope of the patent for the embodiments of the present disclosure should be determined by the appended claims.

Claims

1. A filter parameter determination method, characterized by, For use in an audio system, the audio system including a speaker array and target filters corresponding to the speakers in the speaker array; the method includes: For each of the aforementioned loudspeakers, multiple impulse responses are acquired, and the zero-point parameters and pole parameters corresponding to the loudspeaker are determined based on the multiple impulse responses. The multiple impulse responses are determined based on the received signals obtained by multiple pickup components receiving the audio signals played by the loudspeakers. The filter coefficients corresponding to each target filter are determined according to the zero-point parameters and pole parameters corresponding to each of the speakers respectively. The filter coefficients are used to filter the input audio signal and output a low-frequency audio signal for playback by the corresponding speaker. The step of determining the zero-point parameters and pole parameters corresponding to the loudspeaker based on the plurality of impact responses includes: clustering the plurality of impact responses to obtain a target impact response; making an all-pole assumption on the target impact response to obtain an all-pole expression; solving the all-pole expression using a preset solution algorithm to obtain the coefficients of the all-pole expression, and using the coefficients of the all-pole expression as the pole parameters; and determining the zero-point parameters based on the pole parameters. The step of determining the zero-point parameters based on the pole parameters includes: for each impact response, determining the Z-domain pole expression corresponding to the impact response based on the pole parameters, converting the Z-domain pole expression into a time-domain pole expression, convolving the time-domain pole expression with the impact response to obtain a time-domain zero-point expression; and for each time-domain zero-point expression, performing a fitting process using a preset linear fitting method to obtain the zero-point parameters corresponding to the time-domain zero-point expressions.

2. The method of claim 1, wherein, The acquisition of multiple impact responses includes: Obtain multiple initial shock responses; The multiple initial impulse responses are low-pass filtered to obtain the multiple impulse responses.

3. The method of claim 2, wherein, The acquisition of multiple initial impact responses includes: Multiple received signals are obtained by the multiple pickup components receiving the audio signal played by the speaker respectively; the audio signal played by the speaker is a frequency sweep signal, and each of the pickup components is arranged in the theoretical three-dimensional listening area corresponding to the speaker array; For each of the received signals, the received signal is convolved with the inverse frequency sweep signal to obtain the initial impulse response corresponding to the received signal, wherein the frequency sweep signal and the inverse frequency sweep signal are inverses of each other.

4. The method according to any one of claims 1 to 3, characterized in that, The step of determining the filter coefficients corresponding to each target filter based on the zero-point parameters and pole parameters corresponding to each of the loudspeakers includes: The first coefficient matrix is ​​determined based on the zero-point parameters corresponding to each of the aforementioned loudspeakers; The second coefficient matrix is ​​determined based on the pole parameters corresponding to each of the aforementioned loudspeakers; Multiply the first coefficient matrix by the second coefficient matrix to obtain the target coefficient matrix; The filter coefficients corresponding to each target filter are determined based on the elements included in the target coefficient matrix.

5. The method according to any one of claims 1 to 3, characterized in that, The audio system further includes a high-pass filter and a low-pass filter, and the method further includes: The original audio signal is subjected to high-pass filtering using the high-pass filter to obtain an initial first audio signal, and the initial first audio signal is then delayed to obtain a first audio signal. The original audio signal is low-pass filtered using the low-pass filter to obtain an initial second audio signal, and each of the target filters is used to filter each of the initial second audio signals to obtain the second audio signal output by each of the target filters. Each of the second audio signals is superimposed on the first audio signal to obtain multiple target audio signals, and each of the target audio signals is output by the corresponding loudspeaker using each of the target filters.

6. A filter parameter determination apparatus characterized by comprising: For use in an audio system, the audio system including a speaker array and target filters corresponding to the speakers in the speaker array; the device includes: The first determining module is used to acquire multiple impulse responses for each of the loudspeakers, and determine the zero-point parameters and pole parameters corresponding to the loudspeakers based on the multiple impulse responses, wherein the multiple impulse responses are determined based on the received signals obtained by multiple pickup components receiving the audio signals played by the loudspeakers; The second determining module is used to determine the filter coefficients corresponding to each of the target filters according to the zero-point parameters and pole parameters corresponding to each of the speakers respectively. The filter coefficients are used to filter the input audio signal and output a low-frequency audio signal for playback by the corresponding speaker. The first determining module is configured to: perform clustering processing on the plurality of impact responses to obtain a target impact response; perform all-pole assumption on the target impact response to obtain an all-pole expression; solve the all-pole expression using a preset solving algorithm to obtain the coefficients of the all-pole expression, and use the coefficients of the all-pole expression as the pole parameters; and determine the zero-point parameters based on the pole parameters. The first determining module is configured to: for each of the impact responses, determine the Z-domain pole expression corresponding to the impact response based on the pole parameters, convert the Z-domain pole expression into a time-domain pole expression, convolve the time-domain pole expression with the impact response to obtain a time-domain zero expression; and for each of the time-domain zero expressions, perform fitting processing using a preset linear fitting method to obtain the zero parameters corresponding to the time-domain zero expression.

7. A computer device comprising a memory, a processor, and a computer program stored on the memory and executable on the processor, characterized in that, When the processor executes the computer program, it implements the steps of the method according to any one of claims 1 to 5.

8. A storage medium having a computer program stored thereon, characterized in that, When the computer program is executed by a processor, it implements the steps of the method according to any one of claims 1 to 5.

9. A computer program product, comprising a computer program, characterized in that, When the computer program is executed by a processor, it implements the steps of the method according to any one of claims 1 to 5.

Images