A debugging method and device for an audio system

Abstract

The present application discloses a method for debugging an audio system, the method comprising: using the spatial position information of the second test audio for sound field uniformity calibration played by the sound amplification device collected by the array microphone, obtaining the energy difference of the second test audio at each set test point in the sound field, and determining the second calibration parameter for sound field uniformity calibration according to the energy difference, wherein the second calibration parameter for sound field uniformity calibration is determined in the following manner: determining the playback gain of the sound amplification device according to the energy difference so that the energy difference of each test point reaches the set energy difference threshold. The present application obtains the calibration parameters for sound field calibration through the test audio collected by the array microphone, avoids manual on-site testing, improves the efficiency of sound quality debugging, and is beneficial to the adaptability of the audio system.

Description

Technical Field

[0001] The present invention relates to the field of audio signal processing, and in particular to a debugging method for an audio system. Background Art

[0002] For an audio system composed of a sound amplifier and a microphone, the sound quality cannot be guaranteed in different environments, and experienced technicians are required to debug the sound quality on site, which is time-consuming and labor-intensive. Summary of the invention

[0003] The present invention provides a debugging method for an audio system to improve the sound quality effect adjustment efficiency of the audio system.

[0004] A first aspect of the present invention provides a debugging method for an audio system, the method comprising:

[0005] The energy difference of the second test audio at each set test point in the sound field is obtained by using the spatial position information of the second test audio played by the sound amplification device for sound field uniformity calibration collected by the array microphone.

[0006] determining a second calibration parameter for sound field uniformity calibration according to the energy difference,

[0007] in,

[0008] The second calibration parameter for sound field uniformity calibration is determined as follows:

[0009] The playback gain of the sound amplification device is determined according to the energy difference, so that the energy difference of each test point reaches a set energy difference threshold.

[0010] As a possible implementation, the sound amplification device includes at least one loudspeaker;

[0011] The second test audio is a white noise signal, and the array microphone collects the second test audio played simultaneously by the speakers in the sound amplification device.

[0012] The spatial position information of the second test audio played by the sound amplification device for sound field uniformity calibration collected by the array microphone is obtained, and the energy difference of the second test audio at each set test point in the sound field is obtained, including:

[0013] Obtain the spatial position information of the array microphone and the spatial position information of each test point,

[0014] Based on the spatial position information of the array microphone and the spatial position information of each test point, the collected second test audio is beamformed, and the energy of the beamformed second test audio at each measurement point is calculated.

[0015] Based on the energy of each measurement point, the energy mean of each measurement point is calculated.

[0016] According to the difference between the energy of each measuring point and the energy mean, the energy difference of each measuring point is determined;

[0017] The step of determining the playback gain of the sound amplification device according to the energy difference so that the energy difference of each test point reaches a set energy difference threshold includes:

[0018] For the energy difference at each measuring point, the playback gain of the loudspeaker corresponding to the measuring point is determined according to the energy difference at the measuring point.

[0019] As a possible implementation, the test points are set in the following manner:

[0020] According to the sound field space size information and the layout information of the speakers, the sound field space is divided according to the number of speakers to obtain the sound field space position information corresponding to each speaker.

[0021] At least one test point is set for the sound field space corresponding to each loudspeaker;

[0022] The method of beamforming the second test audio based on the spatial position information of the array microphone and the spatial position information of each test point, and calculating the energy of the beamformed second test audio at each measurement point includes:

[0023] For each test point,

[0024] Based on the spatial position information of the array microphone and the spatial position information of the test point, the relative distance and relative angle between the array microphone and the test point are determined, where the relative angle is used to characterize the angle between the line segment where the relative distance is located and the normal line of the array microphone.

[0025] The relative distance and relative angle of the test point and the spatial position information of each array element in the array microphone are used to generate beamforming coefficients for characterizing the spatial filtering between the measurement point and each array element.

[0026] The collected second test audio is spatially filtered using the beamforming coefficient of the measurement point, and energy calculation is performed on the signal after the spatial filtering to obtain the energy of the beamformed second test audio at the measurement point.

[0027] As a possible implementation manner, dividing the sound field space according to the number of speakers includes:

[0028] Divide the sound field space equally according to the number of speakers so that the sound field space corresponding to each speaker has the same size;

[0029] The projection point of the at least one test point is located at the center of the projection surface of the sound field space corresponding to the loudspeaker;

[0030] The beamforming coefficients are designed based on the beamforming method of Kaiser window;

[0031] The energy difference is the difference between the energy of the measurement point and the energy mean;

[0032] The step of determining the playback gain of the speaker corresponding to each measuring point according to the energy difference of the measuring point includes:

[0033] For each measurement point,

[0034] Determine whether the energy difference of the measurement point is less than the energy difference threshold.

[0035] In the case where the energy difference is not less than the energy difference threshold, if the energy difference is a positive value, the reduced volume of the speaker is determined to be the energy difference; if the energy difference is a negative value, the increased volume of the speaker is determined to be the absolute value of the energy difference.

[0036] Repeat the step of determining whether the energy difference value of the measuring point is less than the energy difference threshold value until the playback gains of the speakers corresponding to the measuring points whose energy difference values ​​are less than the energy difference threshold value are all determined.

[0037] The second test audio played by the speaker at the determined volume collected by the array microphone is obtained, and the step of performing beamforming on the second test audio based on the spatial position information of the array microphone and the spatial position information of each test point and calculating the energy of the beamformed second test audio at each measurement point is returned to be executed until the energy difference of all test points is less than the energy difference threshold.

[0038] As a possible implementation, the method further includes:

[0039] Based on at least one of the first test audio for frequency response calibration of the sound amplifying device and the third test audio for parameter calibration of the array microphone, respectively acquired by the array microphone and played by the sound amplifying device, at least one of the first calibration parameter for frequency response calibration of the sound amplifying device and the third calibration parameter for parameter calibration of the array microphone is obtained;

[0040] in,

[0041] The first calibration parameter is determined in the following manner: based on the first test audio collected by the array microphone, an equalization curve for characterizing the difference between the frequency response of the first test audio and the set target frequency response is obtained, and a parameter for frequency response equalization is determined based on the equalization curve;

[0042] The second calibration parameter is determined in the following manner: based on the third test audio collected by the array microphone, at least one of the environmental noise parameter, the delay parameter, and the reverberation time parameter is obtained, and based on the obtained parameters, at least one of the noise reduction parameter, the echo cancellation parameter, and the reverberation parameter of the array microphone is determined.

[0043] As a possible implementation manner, the first calibration parameter, the second calibration parameter, and the third calibration parameter are determined in a sequential order, so that the second calibration parameter is determined based on the sound amplification device after frequency response calibration, and the third calibration parameter is determined based on the sound amplification device after frequency response calibration and sound field uniformity calibration;

[0044] The array microphone collects first test audio played by each loudspeaker in the sound amplification device;

[0045] The first test audio is a swept frequency audio;

[0046] The method of obtaining an equalization curve for characterizing the difference between the frequency response of the first test audio and the target frequency response based on the first test audio collected by the array microphone, and determining parameters for frequency response equalization based on the equalization curve includes:

[0047] For the first test audio from each speaker collected by the array microphone,

[0048] Performing Fourier transform on the first test audio of the loudspeaker to obtain a frequency response of the first test audio of the loudspeaker,

[0049] Based on the frequency response of the first test audio and the target frequency response of the speaker, an equalization curve of the speaker is obtained.

[0050] At least one of the center frequency, bandwidth, and gain of each peak filter of the loudspeaker is determined according to the equalization curve of the loudspeaker.

[0051] As a possible implementation manner, the third test audio is a maximum length sequence;

[0052] The array microphone collects the third test audio played by each loudspeaker in the sound amplification device;

[0053] The step of obtaining at least one of an environmental noise parameter, a delay parameter, and a reverberation time parameter based on the third test audio collected by the array microphone includes:

[0054] For the third test audio from each speaker collected by the array microphone,

[0055] Calculate the energy of the silent interval in the collected third test audio of the loudspeaker to obtain the environmental noise parameter of the loudspeaker,

[0056] Calculate the cross-correlation function between the maximum length sequence and the collected third test audio of the loudspeaker to obtain the room impulse response function of the loudspeaker,

[0057] Using the room impulse response function, the sound pressure level attenuation curve of the speaker is obtained.

[0058] According to the sound pressure level attenuation curve, determine the maximum delay parameter and the minimum delay parameter of the speaker.

[0059] According to the sound pressure level attenuation curve, determine the reverberation time parameters of the loudspeaker;

[0060] The step of determining at least one of a noise reduction parameter, an echo cancellation parameter, and a reverberation parameter of the array microphone based on the acquired parameters comprises:

[0061] Calculate the average value of the ambient noise parameters of all speakers and set the noise reduction level parameters according to the average value.

[0062] The minimum delay among the delay parameters of all the speakers is selected, and the delay parameter for echo cancellation is determined according to the minimum delay.

[0063] Calculate the difference between the maximum delay and the minimum delay in the delay parameters of all speakers to obtain the delay difference.

[0064] Select the maximum reverberation time among all the speakers’ reverberation times.

[0065] According to the delay difference and the maximum reverberation time, the filter length parameters used for echo cancellation are determined.

[0066] An average value of the reverberation time of all loudspeakers is calculated, and a dereverberation level parameter for dereverberation is determined based on the average value.

[0067] As a possible implementation manner, the obtaining the sound pressure level attenuation curve of the speaker by using the room impulse response function includes:

[0068] The reverse integration of the square of the room impulse response function in the time domain gives a sound energy attenuation curve that characterizes the change of sound energy attenuation over time.

[0069] Each sound energy attenuation value in the sound energy attenuation curve is converted into a logarithm to obtain a sound pressure level attenuation curve;

[0070] Determining the delay parameter of the loudspeaker according to the sound pressure level attenuation curve includes:

[0071] The time corresponding to the maximum value in the sound pressure level attenuation curve is determined as the delay parameter of the loudspeaker;

[0072] Determining the reverberation time parameter of the loudspeaker according to the sound pressure level attenuation curve includes:

[0073] If the sound pressure level in the sound pressure level attenuation curve attenuates by the set first attenuation threshold, the time required for the pressure level attenuation to reach the first attenuation threshold is used as the reverberation time parameter.

[0074] If the sound pressure level attenuation curve cannot reach the attenuation threshold, the sound pressure level attenuation curve in the interval from the initial sound pressure level attenuated by the set second attenuation threshold to the initial sound pressure level attenuated by the set third attenuation threshold is fitted by least squares to obtain the slope of the fitting straight line, and the ratio of the absolute value of the first attenuation threshold to the slope of the fitting straight line is calculated to obtain the reverberation time parameter.

[0075] A second aspect of the present invention provides a debugging device for an audio system, the device comprising:

[0076] The sound field uniformity calibration parameter determination module is used to obtain the energy difference of the second test audio at each set test point in the sound field by using the spatial position information of the second test audio for sound field uniformity calibration played by the sound amplification device collected by the array microphone,

[0077] determining a second calibration parameter for sound field uniformity calibration according to the energy difference,

[0078] in,

[0079] The second calibration parameter for sound field uniformity calibration is determined as follows:

[0080] The playback gain of the sound amplification device is determined according to the energy difference, so that the energy difference of each test point reaches a set energy difference threshold.

[0081] A third aspect of the present invention provides an audio system, which includes the debugging device.

[0082] A fourth aspect of the present invention provides an array microphone, comprising a memory and a processor, wherein the memory stores a computer program, and the processor is configured to execute the computer program to implement the steps of the debugging method of the audio system.

[0083] The debugging method of the audio system provided in the embodiment of the present application obtains the energy difference of the second test audio at each set test point in the sound field by utilizing the spatial position information of the second test audio played by the sound amplification device for sound field uniformity calibration collected by the array microphone, thereby determining the parameters for sound field uniformity calibration based on the energy difference to perform sound field uniformity calibration, so that no additional measurement microphone is required during the debugging process, non-invasive automatic debugging is realized, on-site manual sound quality effect debugging is avoided, the debugging efficiency of the audio system is improved, and the accuracy of sound quality debugging is improved. BRIEF DESCRIPTION OF THE DRAWINGS

[0084] Figure 1 A schematic diagram of a debugging method for an audio system according to an embodiment of the present application.

[0085] Figure 2 A schematic diagram of a top view of the space where the audio system to be debugged is located in this embodiment.

[0086] Figure 3 A schematic diagram of debugging an audio system to be debugged according to this embodiment.

[0087] Figure 4 The figure is a flow chart of frequency response calibration of a loudspeaker in this embodiment.

[0088] Figure 5 A schematic diagram of a flow chart for performing sound field uniformity calibration in this embodiment.

[0089] Figure 6 A schematic diagram of determining the spatial position information of the measuring point in this embodiment.

[0090] Figure 7 Another schematic diagram for determining measurement point information.

[0091] Figure 8 A schematic diagram of the relative distance and relative angle of the measurement point relative to the array microphone.

[0092] Fig. 9 A schematic diagram of a flow chart for calibrating microphone array parameters in this embodiment.

[0093] Fig.10 A schematic diagram of a debugging device for an audio system according to an embodiment of the present application.

[0094] Fig.11 Another schematic diagram of the debugging device for the audio system according to the embodiment of the present application. DETAILED DESCRIPTION

[0095] In order to make the objectives, technical means and advantages of the present application more clearly understood, the present application is further described in detail below in conjunction with the accompanying drawings.

[0096] The embodiment of the present application utilizes the spatial position information of the audio signal collected by the array microphone in the audio system to obtain the energy difference of each test point in the sound field, and performs sound field uniformity calibration according to the energy difference.

[0097] See also Figure 1 As shown, Figure 1 This is a schematic diagram of a debugging method for an audio system according to an embodiment of the present application. The debugging method includes:

[0098] Step 101, using the spatial position information of the second test audio played by the sound amplification device for sound field uniformity calibration collected by the array microphone, obtain the energy difference of the second test audio at each set test point in the sound field,

[0099] As an example, the second test audio is a white noise signal, and the array microphone collects the second test audio played simultaneously by the speakers in the sound amplification device.

[0100] The energy difference of each test point is obtained as follows:

[0101] Obtain the spatial position information of the array microphone and the spatial position information of each test point,

[0102] Based on the spatial position information of the array microphone and the spatial position information of each test point, the second test audio is beamformed to perform spatial filtering on the collected second test audio, and the energy of the beamformed second test audio at each measurement point is calculated.

[0103] Based on the energy of each measurement point, the energy mean of each measurement point is calculated.

[0104] The energy difference of each measuring point is determined according to the difference between the energy of each measuring point and the energy mean.

[0105] As an example, the beamforming method may adopt a constant width beamforming method.

[0106] Step 102: determining a second calibration parameter for sound field uniformity calibration according to the energy difference.

[0107] As an example, the playback gain of the sound amplification device is determined according to the energy difference, so that the energy difference of each test point reaches a set energy difference threshold.

[0108] As yet another example, when the number of the loudspeaker is one, one measurement point may be set, and the energy difference of the measurement point is the difference between the energy of the measurement point and the energy difference threshold.

[0109] The embodiment of the present application does not require an additional measurement microphone, and only uses the array microphone in the audio system to complete the sound field uniformity calibration, thereby realizing the automation of debugging, reducing the workload during the debugging process, and improving the debugging efficiency.

[0110] Furthermore, the debugging method further comprises:

[0111] Based on at least one of the first test audio for frequency response calibration of the sound amplifying device and the third test audio for parameter calibration of the array microphone, respectively acquired by the array microphone and played by the sound amplifying device, at least one of the first calibration parameter for frequency response calibration of the sound amplifying device and the third calibration parameter for parameter calibration of the array microphone is obtained;

[0112] in,

[0113] The first calibration parameter is determined in the following manner: based on the first test audio collected by the array microphone, an equalization curve for characterizing the difference between the frequency response of the first test audio and the set target frequency response is obtained, and a parameter for frequency response equalization is determined based on the equalization curve;

[0114] The third calibration parameter is determined in the following manner: based on the third test audio collected by the array microphone, at least one of the environmental noise parameter, the delay parameter, and the reverberation time parameter is obtained; and based on the obtained parameters, at least one of the noise reduction parameter, the echo cancellation parameter, and the reverberation parameter of the array microphone is determined.

[0115] The above debugging method can be run on an independent debugging client, such as a computer terminal or a mobile terminal installed with the debugging client, or can be run in an audio system, such as any one of a central control terminal, a sound amplification device, and an array microphone in the audio system. The determined calibration parameters are sent to the calibrated end for calibration, for example, the first calibration parameter and the second calibration parameter are sent to the speaker or the central control terminal in the audio system to perform frequency response calibration and sound field uniformity calibration of the speaker, and the third calibration parameter is sent to the array microphone to perform array microphone parameter calibration.

[0116] The audio system debugging method provided in the embodiment of the present application obtains calibration parameters for sound field calibration through test audio collected by an array microphone, thereby avoiding manual on-site testing, improving the efficiency of sound quality debugging, and facilitating the adaptability of the audio system.

[0117] To facilitate understanding of the embodiments of the present application, the following description is given using an example including multiple speakers and an array microphone. It should be understood that the embodiments of the present application are not limited to this, and any audio system including speakers and an array microphone may be applicable, for example, an array microphone conference intercom system, an array microphone local sound reinforcement system, etc.

[0118] See also Figure 2 As shown, Figure 2 This is a schematic diagram of a top view of the space where the audio system to be debugged in this embodiment is located. The audio system to be debugged includes 4 speakers and an array microphone, wherein the 4 speakers are symmetrically distributed in the sound field space, and the array microphone is located above the sound field space covered by the 4 speakers.

[0119] See also Figure 3 As shown, Figure 3 This is a schematic diagram of debugging the audio system to be debugged in this embodiment. In this embodiment, a sound field calibration is performed to detect the current sound field environment and the acoustic characteristics of the current hardware device, including but not limited to the sound field uniformity, device frequency response, environmental noise, delay, reverberation time, etc. The debugging method first performs frequency response calibration on the speaker, then performs sound field uniformity calibration, and finally performs array microphone calibration. Calibration is performed in the above order, which is conducive to improving debugging efficiency.

[0120] See also Figure 4 As shown, Figure 4 This is a flow chart of the frequency response calibration of the speaker in this embodiment. It includes:

[0121] Step 401, obtaining a first test audio for speaker frequency response calibration collected by an array microphone, where the first test audio is played by a single speaker to be calibrated, that is, the first test audio collected by the array microphone and played by a single speaker to be calibrated.

[0122] As an example, the first test audio is a swept frequency audio signal.

[0123] Step 402: Perform Fourier transform on the collected first test audio, for example, perform fast Fourier transform (FFT) to obtain the frequency response of the speaker to be calibrated.

[0124] Step 403: determining an equalization curve of the loudspeaker to be calibrated based on the obtained frequency response of the loudspeaker to be calibrated and the target frequency response, and determining a first calibration parameter for frequency response equalization calibration based on the frequency response equalization curve.

[0125] As an example, the first calibration parameter includes: at least one of the center frequency, bandwidth, and gain of each peak filter of the speaker to be calibrated,

[0126] Step 404: sending the determined first calibration parameter to the speaker to be calibrated, so that the speaker to be calibrated performs frequency response equalization.

[0127] Step 405, repeatedly executing steps 401 to 404 until each speaker in the audio system to be debugged has been frequency-response calibrated one by one.

[0128] As another implementation, when the audio system includes a central control device, the determined first calibration parameter may be sent to the central control device, so that the central control device performs frequency response equalization on the speaker to be calibrated according to the first calibration parameter.

[0129] In the above frequency response calibration process, the first test audio played by each speaker to be calibrated is the same.

[0130] It should be understood that the above steps 401 to 405 may be executed in the same subject or in different subjects. When executed in different subjects, the subjects may be interconnected.

[0131] Based on the speakers after frequency response calibration, perform sound field uniformity calibration. Figure 5 As shown, Figure 5 A schematic diagram of a process for performing sound field uniformity calibration in this embodiment. It includes:

[0132] Step 501, obtaining the sound field spatial dimension information, the loudspeaker spatial position information, and the array microphone spatial position information,

[0133] Among them, the sound field space size, speaker position, and array microphone position information are three-dimensional space information.

[0134] Step 502, determine the spatial position information of each measurement point according to the spatial size information of the sound field,

[0135] As an example, the spatial position information of the measurement point is determined for each speaker according to the sound field space size information and the spatial position information of each speaker. For example, according to the sound field space size information and the layout information of the speakers, the sound field space is divided according to the number of speakers, and the sound field space position information corresponding to each speaker is obtained. A test point is set for the sound field space corresponding to each speaker, so that the projection point of the test point is located at the center of the projection surface of the sound field space corresponding to the speaker, wherein the projection direction of the test point is the same as the projection direction of the sound field space corresponding to the speaker; for example, the projection surface is the ground, and the height of the test point can be determined according to the average height of the human body.

[0136] See also Figure 6 As shown, Figure 6 A schematic diagram for determining the spatial position information of the measurement point in this embodiment. Based on the projection surface of the sound field space covered by each speaker on the ground, the projection surface is evenly divided according to the distribution of each speaker, so that the evenly divided projection surface areas are equal, and the measurement point is set at the center position of each evenly divided projection surface.

[0137] See also Figure 7 As shown, Figure 7 Another schematic diagram for determining the measurement point information. When the sound field space is non-rectangular, such as a symmetrical trapezoid or a quasi-symmetrical trapezoid in a theater, given that the speakers are symmetrically distributed, the projection surface can be symmetrically divided according to the distribution of the speakers, and the measurement point can be set near the center of each symmetrically divided projection surface.

[0138] It should be understood that the number of test points corresponding to each speaker may be the same or different, and the number of test points corresponding to each speaker may be one or more.

[0139] Step 503: Determine the relative distance and relative angle of the measuring point relative to the array microphone based on the spatial position information of the measuring point and the spatial position information of the array microphone.

[0140] See also Figure 8 As shown, Figure 8 is a schematic diagram of the relative distance and relative angle of the measuring point relative to the array microphone, wherein the relative distance is the distance between the measuring point and the array center of the array microphone, that is, the spatial straight-line distance between the two, and the relative angle of the measuring point relative to the array center is the angle between the line segment where the spatial straight-line distance between the two is located and the normal line of the array microphone, such as Figure 8 In the figure, the normal direction of the array microphone is vertically downward, and the angle between the line segment where the spatial straight-line distance between the two is located and the line segment formed by its projection point at the center of the array.

[0141] Step 504, based on the second test audio for sound field uniformity calibration collected by the array microphone, the relative position of the measurement point relative to the array microphone, and the spatial position information of each array element in the array microphone, the energy of the second test audio at each set test point in the sound field is obtained, wherein the second test audio is played simultaneously by all speakers,

[0142] As an example, the second test audio signal is a white noise signal.

[0143] In view of the fact that the second test audio collected by the array microphone has spatial position information, the spatial position information of the collected second test audio is utilized to perform beamforming on the collected second test audio based on the spatial position information of the array microphone and the spatial position information of each test point. In this way, the energy of the second test audio after beamforming at each measurement point can be calculated.

[0144] As an example, the energy of any test point i is obtained as follows:

[0145] For the test point, the relative position information of the measurement point relative to the array microphone and the spatial position information of each array element in the array microphone are used to generate a beamforming coefficient for characterizing the spatial filtering between the measurement point and each array element. For example, the beamforming coefficient is determined based on a Kaiser window. Specifically,

[0146] Mathematically expressed as:

[0147]

[0148] in, is the spatial filtering weight coefficient between the test point i and the array element at position p in the array microphone, is the distance between the array element and the center point of the array, and its value is less than or equal to W / 2. R is the distance from the test point i to the center point of the array. θ is the angle between the line connecting the test point i and the center point of the array and the normal of the array microphone, that is, the relative angle between the test point i and the array microphone. f is the signal frequency. The spatial filtering weight coefficient corresponding to each array element at the test point i can form a column vector, thereby obtaining the beam coefficient of the test point i. represents the window length, which is determined according to the size of the array microphone, β is a parameter for controlling the window shape, and in this embodiment, for each frequency, β is selected to keep the beam width constant, I0(β) represents the first-order zero-order variable Bessel function with the second independent variable β,

[0149] The first independent variable is The zero-order variable Bessel function of the first kind.

[0150] The collected second test audio signal is spatially filtered using a constant-width beam based on the test point to obtain the second test audio at the spatial position of the test point i, that is, the collected second test audio is weighted summed using the beam coefficient of the test point to obtain the second test audio after spatial filtering at the measurement point, that is, the second test audio after beamforming.

[0151] The energy of the second test audio after spatial filtering at the test point is calculated. For example, the mean of the sum of squares of the corresponding values ​​of the second test audio after spatial filtering at each time point is calculated, and the logarithm of the mean with base 10 is obtained. The logarithm is multiplied by 10 to obtain the energy at the test point i, which is expressed as follows:

[0152]

[0153] Among them, L i is the energy at test point i, x n is the signal amplitude corresponding to the sampling time point n in the second test audio signal after spatial filtering, and N is the total number of sampling time points.

[0154] Step 505, obtaining the mean energy of all measurement points,

[0155] As an example, the average value of the energy of all test points is calculated and expressed mathematically as:

[0156] Among them, I is the total number of test points and A is the mean energy.

[0157] Step 506, obtaining the difference between the energy of each measurement point and the mean energy,

[0158] As an example, the difference between the energy of each measurement point and the mean energy is calculated and expressed mathematically as:

[0159] D i =L i -A

[0160] Among them, D i is the difference between the energy of measurement point i and the mean energy.

[0161] Step 507, determine whether the absolute value of each difference value is less than the set energy difference threshold T,

[0162] If the absolute values ​​of the difference values ​​are all less than the energy difference threshold T, it is determined that the sound field uniformity calibration is completed.

[0163] Otherwise, for each difference value that is not less than the energy difference threshold, the playback gain of the speaker corresponding to the difference value is determined according to the difference value, for example, the volume size. When the difference value is a positive value, the degree of reduction of the playback gain of the speaker is determined to be the difference value. When the difference value is a negative value, the degree of increase of the playback gain of the speaker is determined to be the absolute value of the difference value. The determined playback gain is sent to the speaker or the central control device to adjust the playback gain.

[0164] Step 508 : obtaining the second audio signal played by the loudspeaker according to the determined playback gain and collected by the array microphone, and returning to step 504 , until the absolute values ​​of the difference values ​​are all smaller than the energy difference T.

[0165] It should be understood that the above steps 501 to 508 may be executed in the same subject or in different subjects. When executed in different subjects, the subjects only need to be interconnected.

[0166] Calibrate the microphone array parameters based on the speakers that have been calibrated for frequency response and sound field uniformity. Fig. 9 As shown, Fig. 9 A schematic diagram of a flow chart for performing array microphone parameter calibration in this embodiment. It includes:

[0167] Step 901: obtaining a third test audio for array microphone parameter calibration collected by the array microphone, wherein the third test audio is played by a single speaker.

[0168] As an example, the third test audio is a Maximum Length Sequence (MLS) signal.

[0169] Step 902: Acquire environmental noise based on the collected third test audio.

[0170] As an example, the energy of the silent interval in the third test audio is calculated to obtain the ambient noise of the speaker that plays the collected third test audio, which is expressed by a mathematical formula:

[0171]

[0172] Among them, NOISE i is the ambient noise of the speaker i that plays the collected third test audio, y m is the signal amplitude corresponding to the sampling time point m in the collected third test audio signal, and M is the total number of sampling time points.

[0173] Step 903: based on the collected third test audio, obtain a room impulse response RIR of a speaker that plays the collected third test audio.

[0174] Considering that the autocorrelation function of the MLS signal can be approximated as a unit impulse function, the mathematical formula is:

[0175]

[0176] Where S=2 b -1, b is the order, and S is the sequence length of the MLS signal.

[0177] Using the above characteristics of the MLS signal, when the input is an MLS signal, the cross-correlation function between the output and the input is the room impulse response RIR. Therefore, the cross-correlation function between the MLS signal and the collected third test audio is calculated to obtain the room impulse response RIR, which is expressed as follows:

[0178] RIR i =R os (t)

[0179] Among them, R os (t) is the cross-correlation function between the MLS signal and the collected third test audio, RIR i is the room impulse response RIR of the loudspeaker i that plays the collected third test audio.

[0180] Step 904: based on the room impulse response RIR, obtain a sound pressure level attenuation curve of a loudspeaker that plays the collected third test audio.

[0181] As an example, the square of the room impulse response function is reversely integrated in the time domain to obtain a sound energy attenuation curve used to characterize the change of sound energy attenuation over time, which is mathematically expressed as:

[0182]

[0183] In the formula, is the sound energy attenuation curve, is the room impulse response function, and t is the upper limit of time integration.

[0184] The sound pressure level attenuation curve can be obtained by taking the logarithm of the sound energy attenuation curve, which can be expressed mathematically as follows:

[0185] SPL(t)=10lgE(t)

[0186] Among them, SPT is the sound pressure level attenuation value.

[0187] Step 905: based on the sound pressure level attenuation curve, obtain the time delay of the speaker that plays the collected third test audio.

[0188] As an example, the time corresponding to the maximum value in the sound pressure level attenuation curve is taken as the time delay.

[0189] Step 906: based on the sound pressure level attenuation curve, obtain the reverberation time of the speaker that plays the collected third test audio.

[0190] As an example, if the sound pressure level in the sound pressure level attenuation curve attenuates by a set first attenuation threshold, for example, 60 dB, the time required for the pressure level attenuation to reach the first attenuation threshold is used as the reverberation time parameter.

[0191] If the sound pressure level attenuation curve cannot reach the first attenuation threshold, an interval sound pressure level attenuation curve is extracted from the sound pressure level attenuation curve, and the interval sound pressure level attenuation curve is: a curve from the initial sound pressure level attenuated by the sound pressure level attenuation value where the second attenuation threshold is set to the sound pressure level attenuation value where the third attenuation threshold is set, wherein the second attenuation threshold is less than the third attenuation threshold, for example, the second attenuation threshold is 5 dB, and the third attenuation threshold is 25 dB; the least squares linear fitting is performed on the extracted interval sound pressure level attenuation curve to obtain the slope of the fitting straight line, and the ratio of the absolute value of the first attenuation threshold to the slope of the fitting straight line is calculated to obtain the reverberation time parameter, which is expressed by a mathematical formula as follows:

[0192]

[0193] in, is the first attenuation threshold, k is the slope of the fitting line, RT i is the reverberation time of speaker i that plays the collected third test audio.

[0194] Step 907, repeatedly perform steps 901 to 906 until all speakers are tested.

[0195] Step 908, calculate the average environmental noise according to the environmental noise of each speaker, determine the noise reduction level parameter of the array microphone according to the average environmental noise, and send the determined noise reduction level parameter of the array microphone to the array microphone. The calculation of the average environmental noise can be expressed by the mathematical formula:

[0196]

[0197] Where I is the total number of speakers, NOISE i is the ambient noise of speaker i.

[0198] Step 909: taking the minimum value of all speaker delays as the minimum system delay, determining a delay parameter for echo cancellation of the microphone array according to the minimum value, and sending the determined delay parameter to the microphone array.

[0199] Step 910, calculate the difference between the maximum delay and the minimum delay in the delay parameters of all speakers to obtain the delay difference, select the maximum reverberation time in the reverberation time of all speakers, and determine the filter length parameter of the array microphone echo cancellation according to the delay difference and the maximum reverberation time, and send the determined filter length parameter to the array microphone.

[0200] Step 911, calculate the average reverberation time according to the reverberation time of each speaker, determine the dereverberation level parameter of the array microphone according to the average reverberation time, and send the determined dereverberation level parameter to the array microphone. The calculation of the average reverberation time can be expressed by the mathematical formula:

[0201]

[0202] in, is the average reverberation time, I is the total number of loudspeakers, RT i is the reverberation time of speaker i.

[0203] It should be understood that the above steps 901 to 911 can be executed in the same subject or in different subjects. When executed in different subjects, the subjects can be interconnected. Preferably, they can be executed in the array microphone body to improve the efficiency of array microphone parameter calibration.

[0204] This embodiment uses the test audio collected by the array microphone to perform sound field calibration, which not only improves the sound quality of the audio system in different environments, but also realizes automation and simple operation, greatly reduces the difficulty and workload of technical support, and can improve the accuracy of debugging.

[0205] See also Fig.10 As shown, Fig.10This is a schematic diagram of a debugging device for an audio system according to an embodiment of the present application. The device includes:

[0206] The sound field uniformity calibration parameter determination module is used to obtain the energy difference of the second test audio at each set test point in the sound field by using the spatial position information of the second test audio for sound field uniformity calibration played by the sound amplification device collected by the array microphone,

[0207] determining a second calibration parameter for sound field uniformity calibration according to the energy difference,

[0208] in,

[0209] The second calibration parameter for sound field uniformity calibration is determined as follows:

[0210] The playback gain of the sound amplification device is determined according to the energy difference, so that the energy difference of each test point reaches a set energy difference threshold.

[0211] The device also includes:

[0212] The frequency response calibration parameter determination module is used to obtain an equalization curve for characterizing the difference between the frequency response of the first test audio and the set target frequency response based on the first test audio collected by the array microphone, and determine parameters for frequency response equalization based on the equalization curve.

[0213] The array microphone calibration parameter determination module obtains at least one of the environmental noise parameter, the delay parameter, and the reverberation time parameter based on the third test audio collected by the array microphone, and determines at least one of the noise reduction parameter, the echo cancellation parameter, and the reverberation parameter of the array microphone based on the obtained parameters.

[0214] See also Fig.11 As shown, Fig.11 Another schematic diagram of the debugging device of the audio system of the embodiment of the present application. The device includes a memory and a processor, the memory stores a computer program, and the processor is configured to execute the computer program to implement the steps of the debugging method of the audio system of the embodiment of the present application.

[0215] The memory may include a random access memory (RAM) or a non-volatile memory (NVM), such as at least one disk memory. Optionally, the memory may also be at least one storage device located away from the aforementioned processor.

[0216] The above-mentioned processor can be a general-purpose processor, including a central processing unit (CPU), a network processor (NP), etc.; it can also be a digital signal processor (DSP), an application specific integrated circuit (ASIC), a field programmable gate array (FPGA) or other programmable logic devices, discrete gate or transistor logic devices, discrete hardware components.

[0217] An embodiment of the present invention further provides a computer-readable storage medium, in which a computer program is stored. When the computer program is executed by a processor, the steps of the audio system debugging method described in the embodiment of the present application are implemented.

[0218] As for the apparatus / network-side device / storage medium embodiment, since it is basically similar to the method embodiment, the description is relatively simple, and the relevant parts can be referred to the partial description of the method embodiment.

[0219] In this article, relational terms such as first and second, etc. are used only to distinguish one entity or operation from another entity or operation, and do not necessarily require or imply any such actual relationship or order between these entities or operations. Moreover, the terms "comprise", "include" or any other variants thereof are intended to cover non-exclusive inclusion, so that a process, method, article or device including a series of elements includes not only those elements, but also other elements not explicitly listed, or also includes elements inherent to such process, method, article or device. In the absence of further restrictions, the elements defined by the statement "comprise a ..." do not exclude the presence of other identical elements in the process, method, article or device including the elements.

[0220] The above description is only a preferred embodiment of the present invention and is not intended to limit the present invention. Any modifications, equivalent substitutions, improvements, etc. made within the spirit and principles of the present invention should be included in the scope of protection of the present invention.

Claims

1. A method for debugging an audio system, characterized in that: The method includes: The energy difference of the second test audio at each set test point in the sound field is obtained by using the spatial position information of the second test audio played by the sound amplification device for sound field uniformity calibration collected by the array microphone. Determine the second calibration parameter for sound field uniformity calibration according to the energy difference of each test point, in, The spatial position information of the second test audio played by the sound amplification device for sound field uniformity calibration collected by the array microphone is obtained, and the energy difference of the second test audio at each set test point in the sound field is obtained, including: Obtain the spatial position information of the array microphone and the spatial position information of each test point, Based on the spatial position information of the array microphone and the spatial position information of each test point, the collected second test audio is beamformed, and the energy of the beamformed second test audio at each measurement point is calculated. Based on the energy of each measurement point, the energy mean of each measurement point is calculated. According to the difference between the energy of each measuring point and the energy mean, the energy difference of each measuring point is determined; The second calibration parameter for sound field uniformity calibration is determined in the following manner: The playback gain of the sound amplification device is determined according to the energy difference of each measuring point, so that the energy difference of each measuring point reaches a set energy difference threshold.

2. The debugging method according to claim 1, characterized in that: The sound amplification device comprises at least one loudspeaker; The second test audio is a white noise signal; The array microphone collects the second test audio played simultaneously by the loudspeakers in the sound amplification device; The step of determining the playback gain of the sound amplification device according to the energy difference of each measuring point so that the energy difference of each measuring point reaches a set energy difference threshold comprises: For the energy difference at each measuring point, the playback gain of the loudspeaker corresponding to the measuring point is determined according to the energy difference at the measuring point.

3. The debugging method according to claim 2, characterized in that: The test points are set as follows: According to the sound field space size information and the layout information of the speakers, the sound field space is divided according to the number of speakers to obtain the sound field space position information corresponding to each speaker. At least one test point is set for the sound field space corresponding to each loudspeaker; The method of beamforming the second test audio based on the spatial position information of the array microphone and the spatial position information of each test point, and calculating the energy of the beamformed second test audio at each measurement point includes: For each test point, Based on the spatial position information of the array microphone and the spatial position information of the test point, the relative distance and relative angle between the array microphone and the test point are determined, where the relative angle is used to characterize the angle between the line segment where the relative distance is located and the normal line of the array microphone. The relative distance and relative angle of the test point and the spatial position information of each array element in the array microphone are used to generate beamforming coefficients for characterizing the spatial filtering between the measurement point and each array element. The collected second test audio is spatially filtered using the beamforming coefficient of the measurement point, and energy calculation is performed on the signal after the spatial filtering to obtain the energy of the beamformed second test audio at the measurement point.

4. The debugging method according to claim 3, characterized in that: The step of dividing the sound field space according to the number of loudspeakers includes: Divide the sound field space equally according to the number of speakers so that the sound field space corresponding to each speaker has the same size; The projection point of the at least one test point is located at the center of the projection surface of the sound field space corresponding to the loudspeaker; The beamforming coefficients are designed based on the beamforming method of Kaiser window; The energy difference is the difference between the energy of the measurement point and the energy mean; The step of determining the playback gain of the speaker corresponding to each measuring point according to the energy difference of the measuring point includes: For each measurement point, Determine whether the energy difference of the measurement point is less than the energy difference threshold. In the case where the energy difference is not less than the energy difference threshold, if the energy difference is a positive value, the reduced volume of the speaker is determined to be the energy difference; if the energy difference is a negative value, the increased volume of the speaker is determined to be the absolute value of the energy difference. Repeat the step of determining whether the energy difference value of the measuring point is less than the energy difference threshold value until the playback gains of the speakers corresponding to the measuring points whose energy difference values ​​are less than the energy difference threshold value are all determined. The second test audio played by the speaker at the determined volume collected by the array microphone is obtained, and the step of beamforming the second test audio based on the spatial position information of the array microphone and the spatial position information of each test point and calculating the energy of the beamformed second test audio at each measurement point is returned to be executed until the energy difference of all test points is less than the energy difference threshold.

5. The debugging method according to claim 1, characterized in that: The method further comprises: Based on at least one of the first test audio for frequency response calibration of the sound amplifying device and the third test audio for parameter calibration of the array microphone, respectively acquired by the array microphone and played by the sound amplifying device, at least one of the first calibration parameter for frequency response calibration of the sound amplifying device and the third calibration parameter for parameter calibration of the array microphone is obtained; in, The first calibration parameter is determined in the following manner: based on the first test audio collected by the array microphone, an equalization curve for characterizing the difference between the frequency response of the first test audio and the set target frequency response is obtained, and a parameter for frequency response equalization is determined based on the equalization curve; The second calibration parameter is determined in the following manner: based on the third test audio collected by the array microphone, at least one of the environmental noise parameter, the delay parameter, and the reverberation time parameter is obtained, and based on the obtained parameters, at least one of the noise reduction parameter, the echo cancellation parameter, and the reverberation parameter of the array microphone is determined.

6. The debugging method according to claim 5, characterized in that: The first calibration parameter, the second calibration parameter, and the third calibration parameter are determined in a sequential order, such that the second calibration parameter is determined based on the sound amplification device after frequency response calibration, and the third calibration parameter is determined based on the sound amplification device after frequency response calibration and sound field uniformity calibration; The array microphone collects first test audio played by each loudspeaker in the sound amplification device; The first test audio is a swept frequency audio; The method of obtaining an equalization curve for characterizing the difference between the frequency response of the first test audio and the target frequency response based on the first test audio collected by the array microphone, and determining parameters for frequency response equalization based on the equalization curve includes: For the first test audio from each speaker collected by the array microphone, Performing Fourier transform on the first test audio of the loudspeaker to obtain a frequency response of the first test audio of the loudspeaker, Based on the frequency response of the first test audio and the target frequency response of the speaker, an equalization curve of the speaker is obtained. At least one of the center frequency, bandwidth, and gain of each peak filter of the loudspeaker is determined according to the equalization curve of the loudspeaker.

7. The debugging method according to claim 5 or 6, characterized in that: The third test audio is a maximum length sequence; The array microphone collects the third test audio played by each loudspeaker in the sound amplification device; The step of obtaining at least one of an environmental noise parameter, a delay parameter, and a reverberation time parameter based on the third test audio collected by the array microphone includes: For the third test audio from each speaker collected by the array microphone, Calculate the energy of the silent interval in the collected third test audio of the loudspeaker to obtain the environmental noise parameter of the loudspeaker, Calculate the cross-correlation function between the maximum length sequence and the collected third test audio of the loudspeaker to obtain the room impulse response function of the loudspeaker, Using the room impulse response function, the sound pressure level attenuation curve of the speaker is obtained. According to the sound pressure level attenuation curve, determine the maximum delay parameter and the minimum delay parameter of the speaker. According to the sound pressure level attenuation curve, determine the reverberation time parameters of the loudspeaker; The step of determining at least one of a noise reduction parameter, an echo cancellation parameter, and a reverberation parameter of the array microphone based on the acquired parameters comprises: Calculate the average value of the ambient noise parameters of all speakers and set the noise reduction level parameters according to the average value. The minimum delay among the delay parameters of all the speakers is selected, and the delay parameter for echo cancellation is determined according to the minimum delay. Calculate the difference between the maximum delay and the minimum delay in the delay parameters of all speakers to obtain the delay difference. Select the maximum reverberation time among all the speakers’ reverberation times. According to the delay difference and the maximum reverberation time, the filter length parameters used for echo cancellation are determined. An average value of the reverberation time of all loudspeakers is calculated, and a dereverberation level parameter for dereverberation is determined based on the average value.

8. The debugging method according to claim 7, characterized in that: The step of obtaining the sound pressure level attenuation curve of the loudspeaker by using the room impulse response function comprises: The reverse integration of the square of the room impulse response function in the time domain gives a sound energy attenuation curve that characterizes the change of sound energy attenuation over time. Each sound energy attenuation value in the sound energy attenuation curve is converted into a logarithm to obtain a sound pressure level attenuation curve; Determining the delay parameter of the loudspeaker according to the sound pressure level attenuation curve includes: The time corresponding to the maximum value in the sound pressure level attenuation curve is determined as the delay parameter of the loudspeaker; Determining the reverberation time parameter of the loudspeaker according to the sound pressure level attenuation curve includes: If the sound pressure level in the sound pressure level attenuation curve attenuates by the set first attenuation threshold, the time required for the pressure level attenuation to reach the first attenuation threshold is used as the reverberation time parameter. If the sound pressure level attenuation curve cannot reach the attenuation threshold, the sound pressure level attenuation curve in the interval from the initial sound pressure level attenuated by the set second attenuation threshold to the initial sound pressure level attenuated by the set third attenuation threshold is fitted by least squares to obtain the slope of the fitting straight line, and the ratio of the absolute value of the first attenuation threshold to the slope of the fitting straight line is calculated to obtain the reverberation time parameter.

9. A debugging device for an audio system, characterized in that: The device includes: The sound field uniformity calibration parameter determination module is used to obtain the energy difference of the second test audio at each set test point in the sound field by using the spatial position information of the second test audio for sound field uniformity calibration played by the sound amplification device collected by the array microphone, Determine the second calibration parameter for sound field uniformity calibration according to the energy difference of each test point, in, The spatial position information of the second test audio played by the sound amplification device for sound field uniformity calibration collected by the array microphone is obtained, and the energy difference of the second test audio at each set test point in the sound field is obtained, including: Obtain the spatial position information of the array microphone and the spatial position information of each test point, Based on the spatial position information of the array microphone and the spatial position information of each test point, the collected second test audio is beamformed, and the energy of the beamformed second test audio at each measurement point is calculated. Based on the energy of each measurement point, the energy mean of each measurement point is calculated. According to the difference between the energy of each measuring point and the energy mean, the energy difference of each measuring point is determined; The second calibration parameter for sound field uniformity calibration is determined in the following manner: The playback gain of the sound amplification device is determined according to the energy difference of each test point, so that the energy difference of each test point reaches a set energy difference threshold.

10. An audio system, characterized in that: The system comprises the debugging device as claimed in claim 9.

11. An array microphone, characterized in that: The method comprises a memory and a processor, wherein the memory stores a computer program, and the processor is configured to execute the computer program to implement the steps of the debugging method of the audio system according to any one of claims 1 to 8.

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