Sound system automatic debugging method and device and computer equipment

Abstract

The invention relates to a sound system automatic debugging method and device and computer equipment. According to the method, the target curve and the first loss function are set, and the frequency response curve approaches the target curve under multiple influences of the filter, the frequency division point and the gain under multiple iterative operations, so that rapid automatic debugging of the sound system is realized, the operation time of a user is saved, and the learning cost of the user is also reduced.

Description

Technical Field

[0001] The present application relates to the technical field of audio equipment, and in particular to an automatic debugging method, device and computer equipment for an audio system. Background Art

[0002] At present, the conventional debugging method for the sound system composed of active speakers is as follows: (1) first adjust the input processing parameters and output processing parameters of the digital signal processor of a single speaker, such as equalizer, delay, gain, frequency division, etc.; (2) test the effect of the sound system, and adjust the parameters of the digital signal processor of the relevant speakers based on experience and further refinement of the test data; (3) test the overall effect of the sound system again, and adjust the parameters of the digital signal processor of the active speakers again based on the test results. In other words, in order to achieve a better system acoustic effect, it is necessary to repeatedly adjust the parameters of the digital signal processor of each speaker based on the overall acoustic effect.

[0003] This audio system debugging method has the following problems: since there are many factors that affect the acoustic effect of the system, there is no fixed rule as to which speakers to adjust and which parameters of the speakers to adjust, so adjustments can only be made one speaker at a time or one area at a time, and then each speaker must be readjusted according to the test results. Therefore, the above debugging operations are cumbersome, complicated, labor-intensive, and require a high level of professional knowledge reserves from the user. Summary of the invention

[0004] Based on this, it is necessary to provide a method, device and computer equipment for automatic debugging of the audio system to address the above technical problems, so as to realize automatic debugging of the speaker system, replace manual operation, and reduce user time cost and technical threshold.

[0005] In a first aspect, the present application provides a method for automatically debugging a sound system. The method comprises:

[0006] Obtain the actual frequency response curve of the site where the sound system is located and obtain the target curve;

[0007] Obtain a first loss function between the frequency response curve and the target curve under different filters, and determine the target filter according to the first loss function;

[0008] Determine the current target frequency division point based on the target filter, and enter the iteration process, the iteration process includes: taking the gain as a variable, determining the current target gain based on the current target frequency division point; taking the frequency division point as a variable, determining the current target frequency division point based on the current target gain;

[0009] When the preset condition is reached, the iteration is stopped, and the current target frequency division point when the iteration is stopped is obtained as the global target frequency division point, and the current target gain when the iteration is stopped is obtained as the global target gain;

[0010] The sound system can be debugged according to the target filter, global target crossover point and global target gain.

[0011] In one embodiment, determining the current target gain based on the current target frequency division point includes:

[0012] Based on the current target crossover point, the gain is adjusted with a first preset gain interval as a step size, and a gradient algorithm is used to obtain the current target gain;

[0013] Determining the current target crossover point based on the current target gain includes:

[0014] Based on the current target gain, the crossover point is adjusted with a preset frequency interval as the step size, and the current target crossover point is obtained using a gradient algorithm.

[0015] In one embodiment, after determining the target filter, the global target crossover point and the global target gain, the method further includes: automatically debugging the equalizer parameters in the sound system; wherein the equalizer parameters include the center frequency, the Q value and the equalization gain;

[0016] Automatic tuning of equalizer parameters in the sound system includes:

[0017] Find the corresponding frequencies of the areas deviating from the target curve from the frequency response curve and obtain several center frequencies;

[0018] For different center frequencies, target Q values ​​and target equalization gains are obtained.

[0019] In one embodiment, searching the frequency corresponding to the region deviating from the target curve from the frequency response curve to obtain a plurality of center frequencies includes:

[0020] The equalization gain is adjusted with a second preset gain interval as a step length, and the target curve is adjusted based on the equalization gain to obtain a left intersection frequency and a right intersection frequency corresponding to the intersection of the target curve and the frequency response curve under different equalization gains;

[0021] According to each group of left-intersection frequencies and right-intersection frequencies, the corresponding center frequency is determined.

[0022] In one embodiment, for different center frequencies, obtaining a target Q value and a target equalization gain includes:

[0023] For each center frequency, the equalizer parameters are adjusted with a first preset Q value interval as a step length and a third preset gain interval as a step length, and a local equalizer parameter corresponding to the center frequency is obtained according to a second loss function between the frequency response curve and the target curve under different equalizer parameters;

[0024] Comparing the second loss function between the frequency response curve after the local equalizer parameters are adjusted and the target curve, screening the center frequencies corresponding to the local equalizer parameters, and obtaining several target center frequencies;

[0025] For each target center frequency, the equalizer parameters are adjusted with the second preset Q value interval as the step size and the fourth preset gain interval as the step size, and the target Q value and target equalization gain corresponding to the target center frequency are obtained according to the second loss function between the frequency response curve and the target curve under different equalizer parameters.

[0026] In one embodiment, after obtaining the target Q value and the target equalization gain, the method further includes:

[0027] Determine an evaluation value according to a target Q value and a target equalization gain; the evaluation value is the product of the square of the target Q value and the target equalization gain;

[0028] The evaluation value is compared with the first threshold and the second threshold. If the evaluation value is not greater than the first threshold, the target Q value and the target equalization gain corresponding equalizer parameters are retained; if the evaluation value is greater than the first threshold and not greater than the second threshold, the target equalization gain is reduced by one, and the target Q value and the adjusted target equalization gain corresponding equalizer parameters are retained; if the evaluation value is greater than the second threshold, the target Q value and the target equalization gain corresponding equalizer parameters are discarded.

[0029] In one embodiment, after determining the target filter, the global target crossover frequency point and the global target gain, the method further comprises: automatically adjusting the phase in the sound system.

[0030] In a second aspect, the present application also provides a device for automatic debugging of a sound system. The device comprises:

[0031] An acquisition module is used to obtain the actual frequency response curve of the sound system at the site and obtain the target curve;

[0032] A filter selection module is used to obtain a first loss function between a frequency response curve and a target curve under different filters, and determine a target filter according to the first loss function;

[0033] An iterative calculation module is used to determine the current target frequency division point based on the target filter and enter an iterative process, wherein the iterative process includes: taking the gain as a variable and determining the current target gain based on the current target frequency division point; taking the frequency division point as a variable and determining the current target frequency division point based on the current target gain;

[0034] The frequency division module is used to stop the iteration when the preset condition is reached, obtain the current target frequency division point when the iteration stops as the global target frequency division point, and obtain the current target gain when the iteration stops as the global target gain;

[0035] The output module is used to debug the sound system according to the target filter, the global target crossover point and the global target gain.

[0036] In a third aspect, the present application further provides a computer device, which includes a memory and a processor, wherein the memory stores a computer program, and the processor implements the steps in the above-mentioned automatic debugging method of the sound system when executing the computer program.

[0037] In a fourth aspect, the present application further provides a computer-readable storage medium having a computer program stored thereon, which implements the steps in the above-mentioned method for automatically debugging a sound system when the computer program is executed by a processor.

[0038] In a fifth aspect, the present application further provides a computer program product, which includes a computer program, and when the computer program is executed by a processor, the steps in the above-mentioned automatic debugging method of the sound system are implemented.

[0039] The above-mentioned audio system automatic debugging method, device and computer equipment obtain the actual frequency response curve of the site where the audio system is located, obtain the target curve; obtain the first loss function between the frequency response curve and the target curve under different filters, and determine the target filter according to the first loss function; determine the current target crossover point based on the target filter, and enter the iteration process, the iteration process includes: taking the gain as a variable, determining the current target gain based on the current target crossover point; taking the crossover point as a variable, determining the current target crossover point based on the current target gain; stopping the iteration when the preset condition is reached, obtaining the current target crossover point when the iteration stops as the global target crossover point, obtaining the current target gain when the iteration stops as the global target gain; according to the target filter, the global target crossover point and the global target gain, the debugging of the audio system is realized. The present invention sets the target curve and the first loss function, and under multiple iteration operations, the frequency response curve is made to approach the target curve under the multiple influences of the filter, the crossover point and the gain, thereby realizing fast automatic debugging of the audio system, saving the user's operation time, and reducing the user's learning cost. BRIEF DESCRIPTION OF THE DRAWINGS

[0040] Figure 1 A schematic diagram of a flow chart of an automatic debugging method for an audio system in one embodiment;

[0041] Figure 2 is a schematic diagram of a target curve in a sound system in one embodiment;

[0042] Figure 3 A schematic diagram of a frequency division effect in an embodiment;

[0043] Figure 4is a schematic diagram showing the position of the center frequency in one embodiment;

[0044] Figure 5 A schematic diagram showing the comparison before and after phase adjustment in an embodiment;

[0045] Figure 6 It is a structural block diagram of an automatic debugging device for a sound system in one embodiment. DETAILED DESCRIPTION

[0046] In order to make the purpose, technical solution and advantages of the present application more clearly understood, the present application is further described in detail below in conjunction with the accompanying drawings and embodiments. It should be understood that the specific embodiments described herein are only used to explain the present application and are not used to limit the present application.

[0047] The present application embodiment provides a method for automatically debugging an audio system. Figure 1 As shown, the following steps are included:

[0048] Step 102: Obtain an actual frequency response curve of the site where the sound system is located, and obtain a target curve.

[0049] The frequency response curve is a graphical representation of the response of a device to sound signals of different frequencies in an audio system. The frequency response curve is usually measured in the range of 20 Hz to 20 kHz, which is the sound frequency range that the human ear can perceive. Connect the audio signal generator, microphone and audio system under test, use the signal generator to generate a test signal (such as white noise or swept frequency signal) and play it through the audio system under test, the microphone receives the sound signal from the audio system and transmits it to an audio analyzer or computer, and uses audio analysis software to process the received signal to generate a frequency response curve.

[0050] The target curve is a curve chart that has been created and saved. You can freely select or customize the target curve according to the preset scene. Figure 2 Shown is a full-frequency target curve.

[0051] Step 104: Obtain a first loss function between the frequency response curves under different filters and the target curve, and determine the target filter according to the first loss function.

[0052] Among them, the audio system includes various filters, such as high-pass filters, low-pass filters, shelf filters, etc., and LR (Linkwitz-Riley) filters, BW (ButterWorth) filters, BS (Bessel) filters, etc.

[0053] In this embodiment, the user can first set a rough expected crossover point according to the desired crossover effect. For example, for a two-way frequency division, an expected crossover point is set in the middle position to divide the full-frequency frequency response curve into two frequency bands, which requires a high-pass filter and a low-pass filter; for a three-way frequency division, two expected crossover points are set to divide the full-frequency frequency response curve into three frequency bands, which requires a high-pass filter, a low-pass filter and a band-pass filter.

[0054] The filter range is roughly defined according to the expected crossover point, and then the frequency response curve and the target curve of the filters in the corresponding range are processed one by one. The first loss function between the frequency response curve after filter processing and the target curve is calculated, and finally the target filter is determined among various types of filters.

[0055] In this embodiment, M=A-T+F+G is defined, and the first loss function is the sum of the absolute values ​​of each element of the M array, denoted as X, where A represents the frequency curve array, T represents the target curve array, G represents the gain array, and F represents the filter array. The above arrays are all taken from the frequency response diagram, the horizontal axis of the frequency response diagram is the frequency, and the vertical axis is the amplitude. The frequency response diagram is sampled to obtain the frequency curve array, the target curve array, the gain array, and the filter array. It should be noted that in step 104, the gain array is first regarded as a constant, and the filter array is regarded as a variable.

[0056] Through the loop, the first loss function corresponding to all types of filters is calculated, and the filter corresponding to the minimum loss function X is selected as the target filter.

[0057] Different filters have different slopes, such as -6dB, -12dB, -36dB, etc. The slope determines the attenuation degree of the filter to different frequency signals, and is an important parameter for measuring the performance and selectivity of the filter. Therefore, when determining the target filter, it is also necessary to determine the slope corresponding to the filter. This embodiment uses a logarithmic interpolation algorithm to obtain a slope curve. While calculating the first loss function corresponding to all types of filters, the slope curve obtained by the logarithmic interpolation algorithm is also used as a variable to calculate the corresponding first loss function in turn, thereby calculating the slope of the target filter.

[0058] Step 106, determining the current target frequency division point based on the target filter, and entering an iterative process, the iterative process includes: taking the gain as a variable and determining the current target gain based on the current target frequency division point; taking the frequency division point as a variable and determining the current target frequency division point based on the current target gain.

[0059] In step 104, the target filter is determined according to the expected crossover point, and the approximate range of the crossover point is also determined. When the target filter and the filter slope are determined, the first loss function still has two variables, the first is the gain, and the second is the crossover point of the filter.

[0060] At this time, the expected crossover point is used as the current target crossover point of the first iteration. First, the gain is used as a variable to find the gain when the first loss function is the smallest as the current target gain; then the current target gain is used as a constant and the crossover point as a variable to find the crossover point when the first loss function is the smallest as the current target crossover point to complete the first iteration. The current target crossover point output from the first iteration is used as the current target crossover point input for the next iteration.

[0061] Step 108 , when a preset condition is met, the iteration is stopped, and the current target frequency division point when the iteration is stopped is obtained as the global target frequency division point, and the current target gain when the iteration is stopped is obtained as the global target gain.

[0062] Iteration is stopped when the preset condition is reached according to the iteration method of step 104. The preset condition here may refer to the number of iterations, for example, the iteration is stopped when the iteration reaches five or six times. The current target frequency division point when the iteration stops is set as the global target frequency division point, and the current target gain when the iteration stops is set as the global target gain.

[0063] Step 110 , debugging the sound system is performed according to the target filter, the global target crossover point and the global target gain.

[0064] The frequency response curve is processed according to the target filter, the global target crossover point and the global target gain, so that the first loss function between the frequency response curve and the target curve is relatively small, that is, the frequency response curve approaches the target curve, thereby achieving the debugging of the speaker system.

[0065] After the steps disclosed in this embodiment are processed, a better crossover point can be automatically found under the limitation of the expected crossover point, so as to achieve efficient and better sound system debugging. Figure 3 As shown, the set target crossover frequency is 2490 Hz, but the more suitable crossover frequency found by this embodiment is 2676 Hz.

[0066] The user can fill in the parameter target filter, global target crossover point and global target gain into the processor manually or by using the copy function.

[0067] In one embodiment, in step 106, determining the current target gain based on the current target frequency division point includes: adjusting the gain based on the current target frequency division point with a first preset gain interval as a step size, and acquiring the current target gain using a gradient algorithm.

[0068] In a certain iteration process, based on the current target frequency division point, the gain is adjusted starting from 0 with the first preset gain interval as the step length. For example, the first preset gain interval is 0.1. First, the first loss function value X corresponding to 0, 0.1, and -0.1 is calculated. The direction of decreasing values ​​can be determined from the three Xs. The gain is adjusted in the decreasing direction with the first preset gain interval as the step length, and the gain that minimizes X is found as the current target gain corresponding to this iteration.

[0069] In one embodiment, in step 106, determining the current target crossover point based on the current target gain includes: adjusting the crossover point based on the current target gain with a preset frequency interval as a step size, and acquiring the current target crossover point using a gradient algorithm.

[0070] In a certain iteration process, based on the current target gain, the crossover point is adjusted with a preset frequency interval as the step length starting from the current target crossover point outputted in the previous iteration. For example, the preset frequency interval is 1 / 48 oct bandwidth. The minimum loss function value X of each frequency is calculated from the crossover point frequency forward or backward. The X before and after is compared. If it decreases, it means that the gradient decreases. When X starts to increase, the X before the increase is the local minimum first loss function within the current range, and the corresponding crossover point is also the local optimal solution.

[0071] Because the frequency is not high, by verifying multiple times on different types of curves, in this embodiment, it is generally believed that the local optimal solution is the global optimal solution.

[0072] In one embodiment, after determining the target filter, the global target crossover point and the global target gain in step 110, the method further includes: automatically debugging the equalizer parameters in the audio system; wherein the equalizer parameters include the center frequency, the Q value and the equalization gain.

[0073] The center frequency refers to the specific frequency point that the equalizer acts on. The Q value indicates the width of the frequency range that the equalizer acts on. The higher the Q value, the narrower the frequency range; the lower the Q value, the wider the frequency range. The equalization gain refers to the frequency response amplitude that is increased or decreased at a specific frequency.

[0074] Automatically debugging the equalizer parameters in the audio system includes: finding the corresponding frequency of the area deviating from the target curve from the frequency response curve, obtaining several center frequencies; and obtaining the target Q value and target equalization gain for different center frequencies.

[0075] like Figure 4As shown in the figure, the frequency response curve after being processed by the target filter, the global target frequency division point and the global target gain is close to the target curve in general, but it still deviates from the target curve seriously at some frequency points. For these areas, the center frequency is determined. Then, the corresponding equalizer is used for processing according to the different center frequencies, and the target Q value and target equalization gain of the equalizer are determined according to the deviation state between the frequency response curve and the target curve, so that the frequency response curve is close to the target curve. Figure 4 The area circled in the middle is the deviation area, from which the center frequency will be determined.

[0076] In one embodiment, searching the frequency corresponding to the area deviating from the target curve from the frequency response curve to obtain a plurality of center frequencies includes: adjusting the equalization gain with a second preset gain interval as a step size, adjusting the target curve based on the equalization gain, obtaining the left intersection frequency and the right intersection frequency corresponding to the intersection of the target curve and the frequency response curve under different equalization gains; and determining the corresponding center frequency according to each group of left intersection frequencies and right intersection frequencies.

[0077] This embodiment adopts a cyclic algorithm and defines an array N=[A-(T+G)], where A represents the frequency response curve array after being processed by the target filter, the global target frequency division point and the global target gain, T represents the target curve array, and G represents the equalization gain. The equalization gain is adjusted starting from 0 with a second preset gain interval as a step size. For example, the second preset gain interval is set to 0.5, and the array N under different equalization gains such as 0, 0.5, and -0.5 is calculated. According to the change of the array N, it can be judged whether the target curve and the frequency response curve intersect under each equalization gain. For example, when the gain is 0, the corresponding position data of the array N at the 2k frequency is greater than 0, and when the gain is 0.5, the corresponding position data of the array N at the 2k frequency is greater than 0, based on which it can be judged that the target curve intersects with the frequency response curve. Intersecting points often exist in pairs, and are regarded as left intersection points and right intersection points according to the position relationship, and the corresponding frequencies at the intersection are the left intersection frequency and the right intersection frequency.

[0078] The left and right crossover frequencies are averaged and the average is taken as the center frequency.

[0079] In one embodiment, for different center frequencies, obtaining a target Q value and a target equalization gain includes: for each center frequency, adjusting the equalizer parameters with a first preset Q value interval as a step length and a third preset gain interval as a step length, and obtaining local equalizer parameters corresponding to the center frequency according to a second loss function between a frequency response curve and a target curve under different equalizer parameters; comparing the second loss function between the frequency response curve after adjustment of the local equalizer parameters and the target curve, screening the center frequencies corresponding to each local equalizer parameter, and obtaining a number of target center frequencies; for each target center frequency, adjusting the equalizer parameters with a second preset Q value interval as a step length and a fourth preset gain interval as a step length, and obtaining a target Q value and a target equalization gain corresponding to the target center frequency according to the second loss function between the frequency response curve and the target curve under different equalizer parameters.

[0080] First, the center frequency needs to be screened to determine the target center frequency. Specifically, define array Z = AQ-T + F + G, where A is the frequency response curve array, Q is the equalizer array, F is the filter array, G is the gain array, T is the target curve array, and the second loss function is the sum of the absolute values ​​of each element of array Z, which is Y. For each center frequency, adjust the equalizer parameters through loop nesting with the first preset Q value interval as the step size and the third preset gain interval as the step size, calculate the second loss function under different equalizer parameters, and obtain the equalizer parameters when the second loss function is the smallest, which are the local equalizer parameters corresponding to the center frequency. Then take each center frequency and its local equalizer parameters as the equalizer as a whole, and compare the second loss functions between the equalizer as a whole horizontally, so as to screen out several equalizer as a whole with a smaller second loss function, and the corresponding center frequency is the target center frequency.

[0081] After determining the target center frequency, the target Q value and target equalization gain corresponding to each target center frequency are calculated. Specifically, for each target center frequency, the equalizer parameters are adjusted with the second preset Q value interval as the step length and the fourth preset gain interval as the step length, and according to the second loss function between the frequency response curve and the target curve under different equalizer parameters, when the second loss function reaches the minimum, the equalizer parameters corresponding to the target center frequency are obtained, and the target Q value and target equalization gain are analyzed.

[0082] In one embodiment, the first preset Q value interval and the third preset gain interval are relatively large to speed up the cycle speed and improve the screening efficiency of the center frequency; the second preset Q value interval and the fourth preset gain interval are relatively small to determine a more accurate target Q value and target equalization gain. For example, the first preset Q value interval is 0.5, and the Q value range is 0.7 to 4 at this time, and the third preset gain interval is 2, and the equalization gain range is 0 to 10 at this time; the second preset Q value interval is 0.1, and the Q value range is 0.7 to 6 at this time, and the fourth preset gain interval is 0.1, and the equalization gain range is 0 to 10 at this time.

[0083] Usually, the number of equalizers is 12, so 12 target center frequencies can also be selected. After calculating the corresponding equalizer parameters in sequence, 12 equalizers can be obtained.

[0084] In one embodiment, after obtaining the target Q value and the target equalization gain, the method also includes: determining an evaluation value based on the target Q value and the target equalization gain; the evaluation value is the product of the square of the target Q value and the target equalization gain; comparing the evaluation value with a first threshold and a second threshold, if the evaluation value is not greater than the first threshold, retaining the target Q value and the target equalization gain corresponding equalizer parameters; if the evaluation value is greater than the first threshold and not greater than the second threshold, reducing the target equalization gain by one, retaining the target Q value and the adjusted target equalization gain corresponding equalizer parameters; if the evaluation value is greater than the second threshold, discarding the target Q value and the target equalization gain corresponding equalizer parameters.

[0085] Because equalization needs to take human hearing into consideration, the equalizer is screened by Q value and equalization gain. Specifically, the product of the square of the Q value and the equalization gain is calculated as a critical evaluation. When it is less than or equal to the first threshold, it is considered as an equalizer that can be used; when it is greater than the first threshold and less than or equal to the second threshold, the equalization gain is reduced by 1; when it is greater than the second threshold, the equalization is not used. Equalization parameters are provided according to the number of equalizations required by the user.

[0086] In this embodiment, the first threshold value may be 54, and the second threshold value may be 63. The specific values ​​may be adjusted according to actual conditions.

[0087] In one embodiment, after determining the target filter, the global target crossover frequency point and the global target gain, the method further comprises: automatically adjusting the phase in the sound system.

[0088] Phase can be viewed as the relationship between frequency and time. If there is no phase coupling, the phase curves do not overlap. Figure 5 As shown in (a), the boxed area is the area where the phase curves do not overlap.

[0089] The phase loss function is calculated based on the phase data, and the phase array is defined as the phase of the treble - the phase of the bass. The phase loss function is the sum of the absolute values ​​of the elements of the phase array. Among them, the phase of the treble refers to the relative treble and bass formed according to the frequency size after the target crossover point is divided, and are sent to the corresponding speakers respectively. It should be noted that when the phase curve is folded (the phase curve is displayed from -180 degrees to +180 degrees, and when the phase of the two frequencies is greater than 360 degrees, the phase curve is folded), the phase array needs to be flipped by an algorithm, that is, the phase is added by 360, because one fold is equivalent to an extra 360 degrees.

[0090] Use a delay of 0.01ms as a step, loop from -50ms to 50ms, find the delay when the phase loss function is the smallest, then increase the phase of one of the phase curves (treble phase curve or bass phase curve) by 180, that is, perform reverse polarity processing, and then use a delay of 0.01ms as a step, loop from -50ms to 50ms once, and calculate the delay when the phase loss function is the smallest. Compare the two minimum phase loss function values ​​to determine whether the phase curve needs to be reversed; if the minimum phase loss function value with reverse polarity is smaller, the delay obtained after the reverse polarity processing is used as the delay required for the final debugging; otherwise, the delay obtained without the reverse polarity processing is used as the delay required for the final debugging. Adjust the speakers according to the final delay and polarity to allow the phases of different speakers to reach a coupled state. Such as Figure 5 As shown in (b), the boxed area is the area where the phase curves overlap after phase adjustment.

[0091] It should be understood that, although the various steps in the flowcharts involved in the above-mentioned embodiments are displayed in sequence according to the indication of the arrows, these steps are not necessarily executed in sequence according to the order indicated by the arrows. Unless there is a clear explanation in this article, the execution of these steps does not have a strict order restriction, and these steps can be executed in other orders. Moreover, at least a part of the steps in the flowcharts involved in the above-mentioned embodiments can include multiple steps or multiple stages, and these steps or stages are not necessarily executed at the same time, but can be executed at different times, and the execution order of these steps or stages is not necessarily to be carried out in sequence, but can be executed in turn or alternately with other steps or at least a part of the steps or stages in other steps.

[0092] Based on the same inventive concept, the embodiment of the present application also provides an automatic sound system debugging device for implementing the automatic sound system debugging method involved above. The implementation scheme for solving the problem provided by the device is similar to the implementation scheme recorded in the above method, so the specific limitations in one or more embodiments of the automatic sound system debugging device provided below can refer to the limitations of the automatic sound system debugging method above, and will not be repeated here.

[0093] In one embodiment, Figure 6 As shown, a sound system automatic debugging device is provided, comprising:

[0094] The acquisition module 602 is used to obtain the actual frequency response curve of the sound system at the scene and obtain the target curve;

[0095] A filter selection module 604 is used to obtain a first loss function between the frequency response curve and the target curve under different filters, and determine a target filter according to the first loss function;

[0096] The iterative calculation module 606 is used to determine the current target frequency division point based on the target filter and enter an iterative process, wherein the iterative process includes: taking the gain as a variable and determining the current target gain based on the current target frequency division point; taking the frequency division point as a variable and determining the current target frequency division point based on the current target gain;

[0097] The frequency division module 608 is used to stop the iteration when a preset condition is met, obtain the current target frequency division point when the iteration stops as the global target frequency division point, and obtain the current target gain when the iteration stops as the global target gain;

[0098] The output module 610 is used to debug the sound system according to the target filter, the global target crossover point and the global target gain.

[0099] Each module in the above-mentioned automatic debugging device for the sound system can be implemented in whole or in part by software, hardware or a combination thereof. Each of the above-mentioned modules can be embedded in or independent of the processor in the computer device in the form of hardware, or can be stored in the memory in the computer device in the form of software, so that the processor can call and execute the operations corresponding to each of the above modules.

[0100] In one embodiment, a computer device is provided, including a memory and a processor, wherein a computer program is stored in the memory, and the processor implements the steps in all the above method embodiments when executing the computer program.

[0101] In one embodiment, a computer-readable storage medium is provided, on which a computer program is stored. When the computer program is executed by a processor, the steps in all the above method embodiments are implemented.

[0102] In one embodiment, a computer program product is provided, including a computer program, which implements the steps in all the above method embodiments when executed by a processor.

[0103] It should be noted that the user information (including but not limited to user device information, user personal information, etc.) and data (including but not limited to data used for analysis, stored data, displayed data, etc.) involved in this application are all information and data authorized by the user or fully authorized by all parties, and the collection, use and processing of relevant data must comply with relevant laws, regulations and standards of relevant countries and regions.

[0104] Those of ordinary skill in the art can understand that all or part of the processes in the above-mentioned embodiment methods can be completed by instructing the relevant hardware through a computer program, and the computer program can be stored in a non-volatile computer-readable storage medium. When the computer program is executed, it can include the processes of the embodiments of the above-mentioned methods. Among them, any reference to the memory, database or other medium used in the embodiments provided in the present application 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, optical memory, high-density embedded non-volatile memory, resistive random access memory (ReRAM), magnetic random access memory (MRAM), ferroelectric random access memory (FRAM), phase change memory (PCM), graphene memory, etc. Volatile memory can include random access memory (RAM) or external cache memory, etc. As an illustration and not limitation, RAM can be in various forms, such as static random access memory (SRAM) or dynamic random access memory (DRAM). The database involved in each embodiment provided in this application may include at least one of a relational database and a non-relational database. Non-relational databases may include distributed databases based on blockchains, etc., but are not limited thereto. The processors involved in each embodiment provided in this application may be general-purpose processors, central processing units, graphics processors, digital signal processors, programmable logic devices, data processing logic devices based on quantum computing, etc., but are not limited thereto.

[0105] The technical features of the above embodiments may be combined arbitrarily. To make the description concise, 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.

[0106] The above-described embodiments only express several implementation methods of the present application, and the descriptions thereof are relatively specific and detailed, but they cannot be understood as limiting the scope of the present application. It should be pointed out that, for a person of ordinary skill in the art, several variations and improvements can be made without departing from the concept of the present application, and these all belong to the protection scope of the present application. Therefore, the protection scope of the present application shall be subject to the attached claims.

Claims

1. A method for automatic debugging of an audio system, characterized in that: The method comprises: Obtain the actual frequency response curve of the site where the sound system is located and obtain the target curve; Obtaining a first loss function between the frequency response curve and the target curve under different filters, and determining a target filter according to the first loss function; Determine the current target frequency division point based on the target filter, and enter an iterative process, the iterative process comprising: taking the gain as a variable and determining the current target gain based on the current target frequency division point; taking the frequency division point as a variable and determining the current target frequency division point based on the current target gain; When the preset condition is reached, the iteration is stopped, and the current target frequency division point when the iteration is stopped is obtained as the global target frequency division point, and the current target gain when the iteration is stopped is obtained as the global target gain; The sound system is debugged according to the target filter, the global target crossover point and the global target gain.

2. The method according to claim 1, characterized in that Determining the current target gain based on the current target frequency division point includes: Based on the current target frequency division point, the gain is adjusted with a first preset gain interval as a step size, and a gradient algorithm is used to obtain the current target gain; The determining the current target frequency division point based on the current target gain includes: Based on the current target gain, the frequency division point is adjusted with a preset frequency interval as a step size, and the current target frequency division point is acquired by using a gradient algorithm.

3. The method according to claim 1, characterized in that After determining the target filter, the global target crossover point and the global target gain, the method further includes: automatically debugging the equalizer parameters in the sound system; wherein the equalizer parameters include a center frequency, a Q value and an equalization gain; The automatic debugging of the equalizer parameters in the sound system comprises: Find the corresponding frequencies of the regions deviating from the target curve from the frequency response curve to obtain a plurality of center frequencies; For different center frequencies, a target Q value and a target equalization gain are obtained.

4. The method according to claim 3, characterized in that: The step of searching the frequency response curve for the frequency corresponding to the region deviating from the target curve to obtain a plurality of center frequencies comprises: The equalization gain is adjusted with a second preset gain interval as a step size, the target curve is adjusted based on the equalization gain, and a left intersection frequency and a right intersection frequency corresponding to the intersection of the target curve and the frequency response curve under different equalization gains are obtained; According to each group of the left intersection frequency and the right intersection frequency, the corresponding center frequency is determined.

5. The method according to claim 3, characterized in that: The obtaining of the target Q value and the target equalization gain for different center frequencies includes: For each of the center frequencies, the equalizer parameters are adjusted with a first preset Q value interval as a step length and a third preset gain interval as a step length, and a local equalizer parameter corresponding to the center frequency is obtained according to a second loss function between the frequency response curve and the target curve under different equalizer parameters; Comparing the second loss function between the frequency response curve after adjustment of the local equalizer parameters and the target curve, screening the center frequencies corresponding to the local equalizer parameters, and obtaining a plurality of target center frequencies; For each of the target center frequencies, the equalizer parameters are adjusted with a second preset Q value interval as a step size and with a fourth preset gain interval as a step size, and the target Q value and the target equalization gain corresponding to the target center frequency are obtained according to a second loss function between the frequency response curve and the target curve under different equalizer parameters.

6. The method according to claim 3, characterized in that: After obtaining the target Q value and the target equalization gain, the method further includes: Determine an evaluation value according to the target Q value and the target equalization gain; the evaluation value is the product of the square of the target Q value and the target equalization gain; The evaluation value is compared with a first threshold and a second threshold. If the evaluation value is not greater than the first threshold, the target Q value and the target equalization gain corresponding equalizer parameters are retained; if the evaluation value is greater than the first threshold and not greater than the second threshold, the target equalization gain is reduced by one, and the target Q value and the adjusted target equalization gain corresponding equalizer parameters are retained; if the evaluation value is greater than the second threshold, the target Q value and the target equalization gain corresponding equalizer parameters are discarded.

7. The method according to claim 1, characterized in that After determining the target filter, the global target crossover point and the global target gain, the method further includes: automatically adjusting the phase in the sound system.

8. An automatic debugging device for a sound system, characterized in that: The device comprises: An acquisition module is used to obtain the actual frequency response curve of the sound system at the site and obtain the target curve; A filter selection module, used to obtain a first loss function between the frequency response curve and the target curve under different filters, and determine a target filter according to the first loss function; An iterative calculation module is used to determine the current target frequency division point based on the target filter and enter an iterative process, wherein the iterative process includes: taking the gain as a variable and determining the current target gain based on the current target frequency division point; taking the frequency division point as a variable and determining the current target frequency division point based on the current target gain; A frequency division module, used to stop iteration when a preset condition is reached, obtain the current target frequency division point when the iteration stops as the global target frequency division point, and obtain the current target gain when the iteration stops as the global target gain; The output module is used to debug the sound system according to the target filter, the global target frequency division point and the global target gain.

9. A computer device comprising a memory and a processor, wherein the memory stores a computer program, wherein: When the processor executes the computer program, the steps of the method according to any one of claims 1 to 7 are implemented.

10. A computer-readable storage medium having a computer program stored thereon, characterized in that: When the computer program is executed by a processor, the steps of the method according to any one of claims 1 to 7 are implemented.

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