Audio signal adjustment method, device, equipment, medium and program product

Abstract

The invention discloses an audio signal adjustment method and device, equipment, a medium and a program product, and belongs to the field of audio processing. The method is executed by a client, and comprises the following steps: sending a reference audio signal to airborne equipment in a target vehicle; recording to obtain a recorded audio signal, the recorded audio signal being a signal recorded by the client in the cabin of the target vehicle after the airborne equipment plays the reference audio signal; adjusting an audio filter with the recording audio signal and the reference audio signal being the same as each other as a target; and filtering the subsequently played first audio signal through an audio filter to obtain a second audio signal. According to the method, a recorded audio signal is obtained through recording, the recorded audio signal is an audio signal associated with a reference audio signal, and an audio filter is adjusted with the two audio signals being the same as each other, so that a first audio signal played subsequently can be filtered and optimized, and a second audio signal more suitable for being listened in a current cabin is obtained.

Description

Technical Field

[0001] The embodiments of the present application relate to the field of audio processing, and in particular to an audio signal adjustment method, apparatus, device, medium and program product. Background Art

[0002] With the continuous development of the automobile industry, playing music through onboard devices has become a common entertainment item for users while driving.

[0003] In the related art, the onboard device can play the music stored locally, or it can play the music in the terminal device after connecting to the terminal device (such as a smart phone).

[0004] However, due to the special closed scene of the car cabin, unlike the headphone scene or the speaker scene of the terminal device, the music played by the onboard device is usually affected by the car cabin, resulting in poor music quality heard by the user. Summary of the invention

[0005] The present application provides an audio signal adjustment method, apparatus, device, medium and program product, and the technical solution includes at least one of the following aspects.

[0006] According to one aspect of an embodiment of the present application, a method for adjusting an audio signal is provided. The method is performed by a client and includes:

[0007] sending a reference audio signal to an onboard device in a target vehicle;

[0008] Recording to obtain a recorded audio signal, where the recorded audio signal is a signal recorded by the client in the cockpit of the target vehicle after the onboard device plays the reference audio signal;

[0009] Adjust the audio filter with the goal of making the recorded audio signal the same as the reference audio signal;

[0010] The first audio signal played subsequently is filtered through an audio filter to obtain a second audio signal.

[0011] According to another aspect of an embodiment of the present application, there is provided an audio signal adjustment device, the device comprising:

[0012] a transceiver module, for sending a reference audio signal to an onboard device in a target vehicle;

[0013] A recording module, used to record and obtain a recorded audio signal, where the recorded audio signal is a signal recorded by the device in the cockpit of the target vehicle after the onboard device plays the reference audio signal;

[0014] An adjustment module, used for adjusting the audio filter with the goal of making the recorded audio signal and the reference audio signal the same;

[0015] The filtering module is used to filter the first audio signal played subsequently through an audio filter to obtain a second audio signal.

[0016] According to another aspect of an embodiment of the present application, a computer device is provided, comprising: a processor and a memory, wherein at least one program is stored in the memory; the processor is configured to execute at least one program in the memory to implement the above-mentioned audio signal adjustment method.

[0017] According to another aspect of an embodiment of the present application, a computer-readable storage medium is provided, in which at least one program is stored. The at least one program is loaded and executed by a processor to implement the above-mentioned audio signal adjustment method.

[0018] According to another aspect of an embodiment of the present application, a computer program product or a computer program is provided, which includes computer instructions, the computer instructions are stored in a computer-readable storage medium, a processor obtains the computer instructions from the computer-readable storage medium, and the processor executes the computer instructions to implement the above-mentioned audio signal adjustment method.

[0019] The technical solution provided by the embodiments of the present application may have the following beneficial effects:

[0020] The method obtains a recorded audio signal through recording. The recorded audio signal is a signal recorded by a client in the cockpit of a target vehicle after an onboard device plays a reference audio signal. With the goal of making the two audio signals the same, an audio filter is adjusted so that the first audio signal played subsequently can be filtered and optimized through the audio filter to obtain a second audio signal that is more suitable for listening in the current cockpit, thereby improving the listening experience of users in the cockpit. BRIEF DESCRIPTION OF THE DRAWINGS

[0021] In order to more clearly illustrate the technical solutions in the embodiments of the present application, the drawings required for use in the description of the embodiments will be briefly introduced below. Obviously, the drawings described below are only some embodiments of the present application. For ordinary technicians in this field, other drawings can be obtained based on these drawings without paying any creative work.

[0022] Figure 1 A schematic diagram of an audio signal adjustment system provided by an exemplary embodiment of the present application is shown;

[0023] Figure 2 A schematic diagram showing an audio signal adjustment method provided by an exemplary embodiment of the present application is shown;

[0024] Figure 3 A flowchart of an audio signal adjustment method provided by an exemplary embodiment of the present application is shown;

[0025] Figure 4 A flowchart of an audio signal adjustment method provided by an exemplary embodiment of the present application is shown;

[0026] Figure 5 A flowchart of an audio signal adjustment method provided by an exemplary embodiment of the present application is shown;

[0027] Figure 6 A flowchart of an audio signal adjustment method provided by an exemplary embodiment of the present application is shown;

[0028] Figure 7 A flowchart of an audio signal adjustment method provided by an exemplary embodiment of the present application is shown;

[0029] Figure 8 A block diagram of an audio signal adjustment device provided by an exemplary embodiment of the present application is shown;

[0030] Fig. 9 A structural block diagram of a computer device provided by an exemplary embodiment of the present application is shown. DETAILED DESCRIPTION

[0031] In order to make the objectives, technical solutions and advantages of the present application clearer, the implementation methods of the present application will be further described in detail below with reference to the accompanying drawings.

[0032] Exemplary embodiments will be described in detail herein, examples of which are shown in the accompanying drawings. When the following description refers to the drawings, the same numbers in different drawings represent the same or similar elements unless otherwise indicated. The implementations described in the following exemplary embodiments do not represent all implementations consistent with the present application. Instead, they are merely examples of devices and methods consistent with some aspects of the present application as detailed in the appended claims.

[0033] The terms used in this application are for the purpose of describing specific embodiments only and are not intended to limit this application. The singular forms of "a", "said" and "the" used in this application and the appended claims are also intended to include plural forms unless the context clearly indicates other meanings. It should also be understood that the term "and / or" used herein refers to and includes any or all possible combinations of one or more associated listed items.

[0034] It should be noted that the object information (including but not limited to object device information, object 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 object or fully authorized by all parties, and the collection, use and processing of relevant data need to comply with relevant laws, regulations and standards of relevant countries and regions.

[0035] It should be understood that, although the terms first, second, etc. may be used in the present application to describe various information, these information should not be limited to these terms. These terms are only used to distinguish the same type of information from each other. For example, without departing from the scope of the present application, the first parameter may also be referred to as the second parameter, and similarly, the second parameter may also be referred to as the first parameter. Depending on the context, the word "if" as used herein may be interpreted as "at the time of" or "when" or "in response to determining".

[0036] First, some of the terms used in this application are explained:

[0037] Spectrum analysis: It is the process of converting a signal from the time domain to the frequency domain, which can be achieved through Fourier transform (such as fast Fourier transform). After spectrum analysis, the frequency response curve corresponding to the signal can be obtained, which shows the amplitude distribution at different frequencies.

[0038] Frequency response curve: It is a graph that describes the response characteristics of a system or device to signals of different frequencies. The frequency response curve usually uses frequency as the horizontal axis and amplitude (or gain) as the vertical axis, showing the amplification or attenuation of the system or device to signals of different frequencies.

[0039] Reference audio signal: A known signal used for comparison or processing with the actual acquired audio signal. A reference audio signal is a benchmark signal in the audio processing process, used to help improve or analyze the actual acquired audio signal.

[0040] Window function: A mathematical function that calculates the corresponding value through a formula within a specified interval and is zero outside a specified interval. Window functions are used to reduce or control the spectral leakage of signals in frequency domain analysis and signal processing.

[0041] Euler's formula: A formula in complex analysis that establishes the relationship between complex exponential functions and trigonometric functions. The formula is as follows:

[0042] e iθ =cos(θ)+isin(θ);

[0043] Where e is the base of the natural logarithm, i is the imaginary unit, and θ is the angle.

[0044] In signal processing, the spectrum is usually expressed in complex form, where the real part represents the cosine term and the imaginary part represents the sine term. Given the amplitude and phase of a signal, the complex form of the spectrum can be expressed using Euler's formula. Assuming there is a complex number C with amplitude |C| and phase θ, it can be expressed as:

[0045] C=|C|*e iθ ;

[0046] Where e is the base of the natural logarithm and i is the imaginary unit. According to Euler's formula, C can be expressed as:

[0047] C = |C|*(cos(θ)+isin(θ));

[0048] Therefore, the real part Re(C) and imaginary part Im(C) of the complex number C are expressed as:

[0049] Re(C)=|C|*cos(θ);

[0050] Im(C)=|C|*sin(θ);

[0051] Filter: A hardware device or software algorithm used for signal processing. Its function is to extract the required part from the signal while suppressing or removing the unnecessary components. It includes: bandpass filter, high-pass filter, low-pass filter and other types.

[0052] Figure 1 A schematic diagram of an audio signal adjustment system 100 provided by an exemplary embodiment of the present application is shown. The audio signal adjustment system 100 includes: a target vehicle 110 and a terminal device 120 , wherein the target vehicle 110 includes an onboard device 112 .

[0053] In some embodiments, the terminal device 120 includes: a personal computer (PC), a mobile phone, a tablet computer, a wearable device and other electronic devices. The terminal device 120 runs a client 122, and the client 122 has functions such as recording, sending a reference audio signal and controlling the terminal device 120 to play audio. For example, the client 122 is a client corresponding to a music player program. In the embodiments of the present application, the terminal device 120 is generally described as a mobile phone.

[0054] In some embodiments, the target vehicle 110 includes vehicles such as cars (traditional cars, smart cars, unmanned cars, etc.), airplanes, ships, trains, etc. In the embodiments of the present application, the target vehicle 110 is generally described as a car.

[0055] In some embodiments, the target vehicle 110 is equipped with an onboard device 112. The onboard device 112 may be installed with a client running a target application, and the target application may be an application with an audio playback function, such as a music playback program, which is the same as the music playback program corresponding to the client 122. Exemplarily, when the client 122 sends a reference audio signal to the onboard device 112, the onboard device 112 plays the audio corresponding to the reference audio signal.

[0056] With the continuous development of the automobile industry, playing music through onboard devices is a common entertainment item for users while driving. In related technologies, onboard devices can play music stored locally, or they can play music in terminal devices after connecting to terminal devices (such as smartphones). However, due to the special closed scene of the car cabin, unlike the headphone scene or the speaker external scene of the terminal device, the music played by the onboard device is usually affected by the car cabin, resulting in poor music quality heard by users.

[0057] In order to solve the above problems, the present application proposes an audio signal adjustment method. Figure 2 A schematic diagram of an audio signal adjustment method provided by an exemplary embodiment of the present application is shown. The method is executed by a client 122 , and the terminal device 120 runs the client 122 .

[0058] In some embodiments, the client 122 sends a reference audio signal to an onboard device 112 in a target vehicle 110 , and the terminal device 120 is located at a designated position in a cabin of the target vehicle 110 .

[0059] Optionally, the terminal device 120 and the onboard device 112 are connected via Bluetooth.

[0060] Optionally, the client 122 corresponds to the music playing program A, the onboard device 112 runs a car client, and the car client also corresponds to the music playing program A. The client 122 sends the audio signal to the car client, and the car client outputs the audio signal.

[0061] Taking the target vehicle 110 as the target car 110 as an example, in order to improve the quality of the audio heard in the target car 110 , the user can adjust the audio to be played through the audio adjustment function provided by the client 122 .

[0062] Optionally, the user places the terminal device 120 at a designated position in the target car 110 according to actual needs. As an example and not a limitation, when the target car 110 includes only one user sitting in the main driving position, the user places the terminal device 120 in the main driving position; when the target car 110 includes multiple people, the user places the terminal device 120 in the center of the target car 110 or at any seat.

[0063] In some embodiments, the client 122 records and obtains a recorded audio signal, where the recorded audio signal is a signal recorded by the client in the cockpit of the target car after the onboard device 112 plays the reference audio signal.

[0064] Optionally, the client 122 records the audio signal of a preset duration. The preset duration is a value that can be set, for example, 2 seconds or 3 seconds. Too short a recording time will increase the difficulty of adjusting the audio signal, thereby affecting the adjustment effect of the audio signal, and too long a recording time will affect the user's experience.

[0065] In some embodiments, the client 122 performs spectrum analysis on the recorded audio signal to obtain a recorded frequency response curve corresponding to the recorded audio signal.

[0066] Spectral analysis is the process of converting the recorded audio signal from the time domain to the frequency domain, which can be achieved through Fourier transform (such as fast Fourier transform). After spectrum analysis, the recorded frequency response curve corresponding to the recorded audio signal can be obtained, which shows the amplitude distribution at different frequencies.

[0067] Optionally, the client 122 performs spectrum analysis and spectrum smoothing on the recorded audio signal to obtain a recording frequency response curve corresponding to the recorded audio signal.

[0068] Spectral smoothing is a signal processing technique used to reduce noise in spectral data. This is usually achieved through averaging, filtering or other mathematical methods to obtain a smoother frequency response curve.

[0069] In some embodiments, the client 122 performs spectrum analysis on the reference audio signal to obtain a reference frequency response curve corresponding to the reference audio signal.

[0070] like Figure 2 As shown, the solid line is used to represent the reference frequency response curve corresponding to the reference audio signal, and the dotted line is used to represent the recorded frequency response curve corresponding to the recorded audio signal. The recorded frequency response curve and the reference frequency response curve are two curves including the same number of frequency points, and the jth frequency point of the recorded frequency response curve corresponds to the jth frequency point of the reference frequency response curve, where j is an integer less than or equal to the filter order.

[0071] The recorded frequency response curve is different from the reference frequency response curve, that is, the amplitude value corresponding to the j-th frequency point of the recorded frequency response curve is usually different from the amplitude value corresponding to the j-th frequency point of the reference frequency response curve.

[0072] In some embodiments, the audio filter is adjusted with the goal of making the recorded audio signal identical to the reference audio signal.

[0073] Optionally, with the goal of making the recorded frequency response curve identical to the reference frequency response curve, filter coefficients of an audio filter are calculated, and the audio filter is used to adjust the frequency response curve corresponding to the audio signal.

[0074] The filter coefficients of an audio filter are a series of constants used to define the characteristics of an audio filter. Audio filters are subject to the following formula:

[0075]

[0076] Where y(n) is the output signal at time n, x(nk) is the input signal at time nk, and k is an integer from 0 to N-1. h(k) is the filter coefficient of the audio filter, there are N of them, and N is the value of the filter order plus 1. The filter order is related to the frequency response of the audio filter and is a preset value.

[0077] In some embodiments, the frequency points of the recorded frequency response curve correspond to the frequency points of the reference frequency response curve one by one, and the filter coefficients of the audio filter are calculated, including:

[0078] Calculate the amplitude difference between the jth frequency point of the recorded frequency response curve and the jth frequency point of the reference frequency response curve;

[0079] Calculate the phase of each frequency point based on a preset filter order;

[0080] Based on the phase of each frequency point and the difference of the amplitude value corresponding to each frequency point, the imaginary part and the real part of the first spectrum are obtained by Euler's formula, and the second spectrum corresponding to the first spectrum is calculated by conjugation;

[0081] Performing an inverse Fourier transform on each frequency point in the first spectrum and the second spectrum to obtain a first time domain signal corresponding to both the first spectrum and the second spectrum;

[0082] Performing numerical normalization on the first N real parts of the first time domain signal, where N is the value of the filter order plus 1;

[0083] The first N real part data after numerical normalization are processed by a window function to obtain the filter coefficients of the audio filter.

[0084] In some embodiments, the audio filter corresponds to a first group delay, the first group delay is used to represent the delay degree of each frequency component in the audio signal, and the phase of each frequency point is calculated through the first group delay.

[0085] The calculation method of the first group delay GroundDelay is as follows:

[0086] GroundDelay = order / 2;

[0087] Here, order is the filter order.

[0088] The calculation method of the phase θ at the jth frequency point is as follows:

[0089] θ[j]=-GroundDelay*PI*j / length;

[0090] Where j is the serial number of the frequency point, PI is the circumference of a circle, length is the length of the effective spectrum, length = nfft / 2+1, nfft is the number of frequency points associated with the spectrum analysis (Fourier transform), that is, the length of the Fourier transform, and nfft is calculated as follows:

[0091] nfft=2*(pow(2,ceil(log(order+1) / log(2)))-1);

[0092] Among them, the ceil function is a rounding function, and the pow function is a power operation function, which is used here to calculate the ceil power of 2.

[0093] Assume that there is a complex number C with amplitude |C| and phase θ, then it can be expressed as:

[0094] C=|C|*e iθ ;

[0095] Where e is the base of the natural logarithm and i is the imaginary unit. According to Euler's formula, C can be expressed as:

[0096] C = |C|*(cos(θ)+isin(θ));

[0097] Therefore, the real part Re(C) and imaginary part Im(C) of the complex number C are expressed as:

[0098] Re(C)=|C|*cos(θ);

[0099] Im(C)=|C|*sin(θ);

[0100] After obtaining the phase (θ) of each frequency point and the amplitude value difference (C) corresponding to each frequency point, the imaginary part and the real part of the first spectrum can be obtained by using the Euler formula.

[0101] In some embodiments, the N real part data after numerical normalization are processed by a Hanning window function to obtain filter coefficients of the audio filter.

[0102] Optionally, the window function includes at least one of the following: a rectangular window function, a Hanning window function, a Hamming window function, and a Blackman window function. The embodiment of the present application uses the Hanning window function as an example, and does not limit the window function.

[0103] The Hanning window function is expressed by the following formula:

[0104]

[0105] Where N is the length of the Hanning window, and n is the sequence number of the real data.

[0106] Exemplarily, the first filter coefficient is the product of w(0) and the first real data after numerical normalization, the second filter coefficient is the product of w(1) and the second real data after numerical normalization, and the nth filter coefficient is the product of w(n) and the nth real data after numerical normalization.

[0107] In some embodiments, the first audio signal played subsequently is filtered by an audio filter to obtain a second audio signal.

[0108] After the filter coefficient of the audio filter is calculated, the audio filter is applied to filter the first audio signal that is subsequently played, thereby adjusting the first audio signal and obtaining a second audio signal with better quality for the user.

[0109] In some embodiments, the method further comprises: adjusting the second audio signal by using the first algorithm and the second algorithm to obtain an output audio signal;

[0110] The first algorithm is used to improve the stereoscopic effect of the audio corresponding to the output audio signal, and the second algorithm is used to enhance the spatial effect of the audio corresponding to the output audio signal.

[0111] In some embodiments, the second audio signal is adjusted by a first algorithm to obtain a third audio signal; and the third audio signal is adjusted by a second algorithm to obtain an output audio signal.

[0112] First algorithm: Optionally, the first algorithm corresponds to two feasible implementation plans: Plan 1 and Plan 2.

[0113] Solution 1: In some embodiments, first left channel data and first right channel data corresponding to the second audio signal are obtained;

[0114] Calculate first edge data and first intermediate data based on the left channel data and the right channel data;

[0115] Performing weighted calculation on the first edge data and the first intermediate data to obtain second edge data and second intermediate data;

[0116] Based on the second edge data and the second intermediate data, second left channel data and second right channel data are calculated, and the third audio signal corresponds to the second left channel data and the second right channel data.

[0117] Exemplarily, the first left channel data is left, and the first right channel data is right. The first left channel data and the first right channel data can be single data or arrays including multiple data, for example: left = [10, 8, 6, 8, 20, 24], right = [12, 8, 2, 10, 16, 22].

[0118] Assume that the first edge data is sides, the first middle data is mid, sides=(1 / 2)*(left-right); mid=(1 / 2)*(left+right).

[0119] The second edge data is newsides, the second middle data is newmid, the weight coefficient of the weighted calculation association is width, and width is greater than or equal to 0 and less than or equal to 2. newsides=width*sides, newmid=(2-width)*mid.

[0120] Finally, the second left channel data and the second right channel data are calculated. The second left channel data is newleft, and the second right channel data is newright. newleft = newmid + newsides, and the second right channel data newright = newmid - newsides.

[0121] Solution 2: In some embodiments, first left channel data and first right channel data corresponding to the second audio signal are obtained; one of the first left channel data or the first right channel data is delayed for a specified time length, and the other data remains unchanged to obtain a third audio signal.

[0122] Exemplarily, the specified time length is set within 5-35 milliseconds. Assuming that the terminal device 120 is in the center of the target car 110, the specified time length is set to 5 milliseconds; assuming that the terminal device 120 is in the main driving position or the co-pilot position of the target car, the specified time length is set to 20 milliseconds.

[0123] Second algorithm: In some embodiments, adjusting the third audio signal by the second algorithm to obtain the output audio signal includes:

[0124] filtering the third audio signal through a bandpass filter to obtain a fourth audio signal, filtering the third audio signal through a high-pass filter to obtain a fifth audio signal, and filtering the third audio signal through a low-pass filter to obtain a sixth audio signal;

[0125] The fourth audio signal, the fifth audio signal and the sixth audio signal are superimposed to obtain an output audio signal.

[0126] A high-pass filter is a filter that allows high-frequency signals to pass through while blocking low-frequency signals below the cutoff frequency; a low-pass filter is a filter that allows low-frequency signals to pass through while blocking high-frequency signals above the cutoff frequency. A band-pass filter is a filter that allows signals within a specific frequency range to pass through while blocking frequencies outside the specific frequency range. The specific frequency range is called the passband, and the frequency range outside the passband is called the stopband. After applying the band-pass filter, a fourth audio signal containing only specific frequency components can be obtained.

[0127] To sum up, the method provided in this embodiment obtains a recorded audio signal through recording. The recorded audio signal is a signal recorded by the client in the cabin of the target vehicle after the onboard device plays the reference audio signal. With the goal of making the two audio signals the same, the audio filter is adjusted so that the first audio signal played subsequently can be filtered and optimized through the audio filter to obtain a second audio signal that is more suitable for listening in the current cabin, thereby improving the listening experience of the user in the cabin.

[0128] The method provided in this embodiment also adjusts the second audio signal through the first algorithm and the second algorithm to obtain the output audio signal, thereby improving the stereoscopic and spatial sense of the audio corresponding to the output audio signal and reducing the problem of difficulty for users in selecting appropriate sound effects.

[0129] Figure 3 A flowchart of an audio signal adjustment method provided by an exemplary embodiment of the present application is shown. The method is executed by a client and includes at least one of the following steps.

[0130] Step 310: Send a reference audio signal to an onboard device in a target vehicle.

[0131] In some embodiments, the target transportation means include: automobiles (traditional automobiles, smart automobiles, unmanned vehicles, etc.), airplanes, ships, trains, etc. In the embodiments of the present application, the target transportation means is usually described as an example of an automobile.

[0132] In some embodiments, the target vehicle is equipped with an onboard device, which may be a control system of the target vehicle. A client running a target application may be installed on the onboard device, which may be an application with an audio playback function, such as a music playback program.

[0133] Exemplarily, when the client sends a reference audio signal to the onboard device, the onboard device plays the audio corresponding to the reference audio signal.

[0134] In some embodiments, the reference audio signal is a known signal used for comparison or processing with an actually acquired audio signal (eg, a recorded audio signal).

[0135] The reference audio signal is a reference signal in the audio processing process, which is used to help improve or analyze the actually acquired audio signal. For example, the reference audio signal includes a swept frequency signal. A swept frequency signal is a signal whose frequency changes continuously within a certain time range.

[0136] Step 320: Record and obtain a recorded audio signal.

[0137] The recorded audio signal is a signal recorded by the client in the cockpit of the target vehicle after the onboard device plays the reference audio signal.

[0138] In some embodiments, the client runs in the terminal device. The client has functions such as recording, sending reference audio signals, and controlling the terminal device to play audio. In the embodiments of the present application, the terminal device is usually described as a mobile phone.

[0139] Optionally, the user places the terminal device at a designated location in the target vehicle according to actual needs. For example, if the target vehicle is a target car, when the target car includes only one user sitting in the main driving position, the user places the terminal device in the main driving position; when the target car includes multiple people, the user places the terminal device in the center of the target car or at any seat.

[0140] Optionally, a recording audio signal of a preset duration is recorded. The preset duration is a value that can be set, for example, 2 seconds or 3 seconds. Too short a recording time will increase the difficulty of adjusting the audio signal, thereby affecting the adjustment effect of the audio signal. Too long a recording time will affect the user experience.

[0141] Step 330: Adjust the audio filter with the goal of making the recorded audio signal identical to the reference audio signal.

[0142] like Figure 2As shown, the solid line is used to represent the reference frequency response curve corresponding to the reference audio signal, and the dotted line is used to represent the recorded frequency response curve corresponding to the recorded audio signal. The recorded frequency response curve and the reference frequency response curve are two curves including the same number of frequency points, and the jth frequency point of the recorded frequency response curve corresponds to the jth frequency point of the reference frequency response curve, where j is an integer less than or equal to the filter order. The filter order is associated with the frequency response of the audio filter, and the filter order is a preset value.

[0143] The recorded frequency response curve is different from the reference frequency response curve, that is, the amplitude value corresponding to the j-th frequency point of the recorded frequency response curve is usually different from the amplitude value corresponding to the j-th frequency point of the reference frequency response curve.

[0144] Since the car cabin is a closed environment, the audio signal played by the onboard device will be affected by factors such as sound reflection and environmental noise, resulting in the audio signal heard by the user being different from the reference audio signal, that is, the recorded audio signal is different from the reference audio signal. In order to make the recorded audio signal similar to or even the same as the reference audio signal, the audio filter needs to be adjusted, for example, the filter coefficient of the audio filter needs to be adjusted.

[0145] Step 340: filtering the first audio signal to be subsequently played through an audio filter to obtain a second audio signal.

[0146] The second audio signal is an audio signal after filtering and optimizing the first audio signal. The second audio signal is similar to the first audio signal. For the second audio signal, after reflection and refraction in the cabin, when it reaches the location of the passenger terminal device, it is closer to the original audio quality of the first audio signal. If the onboard device directly plays the first audio signal, the audio quality of the audio signal received by the user will be affected by factors such as sound reflection and environmental noise, and will not be close to the original audio quality of the first audio signal.

[0147] After the audio filter is adjusted, the audio filter is applied to filter the first audio signal that is subsequently played, thereby adjusting the first audio signal to obtain a second audio signal with better quality for the user.

[0148] To sum up, the method provided in this embodiment obtains a recorded audio signal through recording. The recorded audio signal is a signal recorded by the client in the cabin of the target vehicle after the onboard device plays the reference audio signal. With the goal of making the two audio signals the same, the audio filter is adjusted so that the first audio signal played subsequently can be filtered and optimized through the audio filter to obtain a second audio signal that is more suitable for listening in the current cabin, thereby improving the listening experience of the user in the cabin.

[0149] Figure 4A flowchart of an audio signal adjustment method provided by an exemplary embodiment of the present application is shown. The method is executed by a client and includes at least one of the following steps.

[0150] Step 310: Send a reference audio signal to an onboard device in a target vehicle.

[0151] Step 320: Record and obtain a recorded audio signal.

[0152] Step 332: Perform spectrum analysis, and obtain a recording frequency response curve corresponding to the recorded audio signal and a reference frequency response curve corresponding to the reference audio signal.

[0153] Spectrum analysis is the process of converting the recorded audio signal and the reference audio signal from the time domain to the frequency domain, which can be achieved through Fourier transform (such as fast Fourier transform). After spectrum analysis, the recorded frequency response curve and the reference frequency response curve can be obtained, which show the amplitude distribution at different frequencies.

[0154] The frequency response curve is a graph that describes the response characteristics of a system or device to signals of different frequencies. The frequency response curve usually uses frequency as the horizontal axis and amplitude (or gain) as the vertical axis, showing the amplification or attenuation of the system or device to signals of different frequencies.

[0155] In some embodiments, spectrum analysis and spectrum smoothing are performed, and a recorded frequency response curve corresponding to the recorded audio signal and a reference frequency response curve corresponding to the reference audio signal are obtained respectively.

[0156] Spectral smoothing is a signal processing technique used to reduce noise in spectral data. It is usually achieved by averaging, filtering or other mathematical methods to obtain a smoother frequency response curve.

[0157] In some embodiments, frequency alignment of the recorded frequency response curve and the reference frequency response curve is performed using an interpolation algorithm.

[0158] By using an interpolation algorithm, the jth frequency point of the recorded frequency response curve is made to correspond to the jth frequency point of the reference frequency response curve, where j is an integer less than or equal to the filter order.

[0159] Exemplarily, the interpolation algorithms include linear interpolation, nearest neighbor interpolation, cubic spline interpolation, cubic spline interpolation and the like, which are not limited in the embodiments of the present application.

[0160] In some embodiments, the degree of curve fitting is controlled by setting an amplitude threshold.

[0161] Exemplarily, the amplitude threshold includes a maximum boost amplitude and a maximum attenuation amplitude. The maximum boost amplitude is the maximum gain or boost degree when adjusting the amplitude value, and the maximum attenuation amplitude is the maximum reduction or attenuation degree when adjusting the amplitude value.

[0162] In some embodiments, after performing spectrum analysis, the system low-frequency characteristics corresponding to the target vehicle are obtained according to the original frequency response curve; a low-frequency enhancement signal corresponding to the system low-frequency characteristics is designed; and the low-frequency enhancement signal is superimposed on the reference frequency response curve corresponding to the reference audio signal.

[0163] Step 334: Calculate the filter coefficients of the audio filter with the goal of making the recorded frequency response curve identical to the reference frequency response curve.

[0164] The audio filter is used to adjust the frequency response curve corresponding to the audio signal.

[0165] The filter coefficients of an audio filter are a series of constants used to define the characteristics of an audio filter. Audio filters are subject to the following formula:

[0166]

[0167] Where y(n) is the output signal at time n, x(nk) is the input signal at time nk, and k is an integer from 0 to N-1. h(k) is the filter coefficient of the audio filter, there are N of them, N is the value of the filter order plus 1, and the filter order is a positive integer. The filter order is associated with the frequency response of the audio filter, and the filter order is a preset value.

[0168] Figure 5 A flowchart of an audio signal adjustment method provided by an exemplary embodiment of the present application is shown. In some embodiments, the filter coefficients of the audio filter are calculated with the goal of making the recorded frequency response curve the same as the reference frequency response curve, including:

[0169] Step 510: Obtain a recording frequency response curve corresponding to the recorded audio signal and a reference frequency response curve corresponding to the reference audio signal.

[0170] Optionally, a method of acquiring the recording frequency response curve and the reference frequency response curve includes at least one of the following: direct input, array pointer, and text file.

[0171] The array pointer is used to store the memory address of the array. By passing the array pointer to the client, the client can directly access and process the data in the array. A text file is a file format that can contain data consisting of characters. The data of the recorded frequency response curve and the reference frequency response curve are stored in the text file in the form of text. Each line or each separated value in the text file can represent a frequency point and the corresponding amplitude. The frequency points in the two frequency response curves are divided in the same way.

[0172] Step 520: Calculate the amplitude difference between the jth frequency point of the recorded frequency response curve and the jth frequency point of the reference frequency response curve.

[0173] With the goal of making the recorded frequency response curve identical to the reference frequency response curve, the amplitude difference between the jth frequency point of the recorded frequency response curve and the jth frequency point of the reference frequency response curve is calculated. The frequency points of the recorded frequency response curve correspond to the frequency points of the reference frequency response curve one by one, and j is an integer less than or equal to the filter order.

[0174] Step 530: Calculate the phase of each frequency point based on a preset filter order.

[0175] In some embodiments, the audio filter corresponds to a first group delay, the phase of each frequency point is calculated by the first group delay, and the value of the first group delay is half of the filter order.

[0176] The first group delay is used to indicate the delay degree of each frequency component in the audio signal. The calculation method of the first group delay GroundDelay is as follows:

[0177] GroundDelay = order / 2;

[0178] Here, order is the filter order.

[0179] In some embodiments, the phase of the jth frequency point is the quotient of the first product and the length of the effective spectrum;

[0180] The first product is the product of the inverse of the first group delay, pi, and j, and the length of the effective spectrum is half the number of frequency points associated with the spectrum analysis plus 1.

[0181] The calculation method of the phase θ at the jth frequency point is as follows:

[0182] θ[j]=-GroundDelay*PI*j / length;

[0183] Where j is the serial number of the frequency point, PI is the circumference of a circle, length is the length of the effective spectrum, length = nfft / 2+1, nfft is the number of frequency points associated with the spectrum analysis (Fourier transform), that is, the length of the Fourier transform, and nfft is calculated as follows:

[0184] nfft=2*(pow(2,ceil(log(order+1) / log(2)))-1);

[0185] Among them, the ceil function is a rounding function, and the pow function is a power operation function, which is used here to calculate the ceil power of 2.

[0186] In some embodiments, the method further includes: performing weighted smoothing on the amplitude value differences corresponding to each frequency point.

[0187] Step 540: Based on the phase of each frequency point and the amplitude value difference corresponding to each frequency point, the imaginary part and the real part of the first spectrum are obtained by Euler's formula, and the second spectrum corresponding to the first spectrum is calculated by conjugation.

[0188] Assume that there is a complex number C with amplitude |C| and phase θ, then it can be expressed as:

[0189] C=|C|*e iθ ;

[0190] Where e is the base of the natural logarithm and i is the imaginary unit. According to Euler's formula, C can be expressed as:

[0191] C = |C|*(cos(θ)+isin(θ));

[0192] Therefore, the real part Re(C) and imaginary part Im(C) of the complex number C are expressed as:

[0193] Re(C)=|C|*cos(θ);

[0194] Im(C)=|C|*sin(θ);

[0195] After obtaining the phase (θ) of each frequency point and the amplitude value difference (C) corresponding to each frequency point, the imaginary part and the real part of the first spectrum can be obtained by using the Euler formula.

[0196] In some embodiments, the imaginary part and the real part of the first spectrum are obtained by Euler's formula, and the second spectrum corresponding to the first spectrum is calculated by conjugation.

[0197] For example, the imaginary part of the first spectrum is |C|*sin(θ), and the real part is |C|*cos(θ), then the imaginary part of the second spectrum is -|C|*sin(θ), and the real part is |C|*cos(θ).

[0198] Step 550: Perform inverse Fourier transform on each frequency point in the first spectrum and the second spectrum to obtain a first time domain signal corresponding to both the first spectrum and the second spectrum.

[0199] The inverse Fourier transform is the process of converting the first spectrum and the second spectrum from the frequency domain to the time domain. After performing the inverse Fourier transform, the complex numbers of each frequency point correspond to the complex numbers of each time point in the first time domain signal, and the real part of the complex number at each time point corresponds to the real part of the complex number at each frequency point.

[0200] Step 560: Perform numerical normalization on the first N real parts in the first time domain signal, where N is the value of the filter order plus 1.

[0201] In some embodiments, numerical normalization is performed by dividing the first N real parts of the first time domain signal by the length of the inverse Fourier transform, respectively, and the length of the inverse Fourier transform is the same as the number of frequency points associated with the spectrum analysis.

[0202] The number of frequency points associated with the spectrum analysis, that is, the length of the Fourier transform (nfft), the calculation method of nfft refers to step 530 and will not be repeated here.

[0203] Step 570: Process the first N real part data after numerical normalization through a window function to obtain filter coefficients of the audio filter.

[0204] Optionally, the window function includes at least one of the following: a rectangular window function, a Hanning window function (Hanning Window), a Hamming window function (Hamming Window), and a Blackman window (Blackman Window).

[0205] In some embodiments, the N real part data after numerical normalization are processed by a Hanning window function to obtain filter coefficients of the audio filter.

[0206] The Hanning window function is expressed by the following formula:

[0207]

[0208] Where N is the length of the Hanning window, and n is the sequence number of the real data.

[0209] Exemplarily, the first filter coefficient is the product of w(0) and the first real data after numerical normalization, the second filter coefficient is the product of w(1) and the second real data after numerical normalization, and the nth filter coefficient is the product of w(n) and the nth real data after numerical normalization.

[0210] Step 340: filtering the first audio signal to be subsequently played through an audio filter to obtain a second audio signal.

[0211] In summary, the method provided in this embodiment performs spectrum analysis and obtains a recorded frequency response curve corresponding to the recorded audio signal and a reference frequency response curve corresponding to the reference audio signal respectively; with the goal of making the recorded frequency response curve identical to the reference frequency response curve, the filter coefficient of the audio filter is calculated, thereby adjusting the audio filter more reasonably and accurately, thereby improving the filtering effect of the audio filter on the first audio signal.

[0212] The method provided in this embodiment also converts the audio signal from the time domain to the frequency domain through spectrum analysis (such as Fourier transform), which is conducive to analyzing the frequency characteristics of the audio signal and facilitating filtering, smoothing and other operations on the audio signal; and converts the first spectrum and the second spectrum from the frequency domain to the time domain through inverse Fourier transform, thereby reconstructing the audio signal.

[0213] The method provided in this embodiment also calculates the amplitude difference between the jth frequency point of the recorded frequency response curve and the jth frequency point of the reference frequency response curve, and calculates the phase of each frequency point based on a preset filter order, thereby accurately calculating each frequency point corresponding to the first spectrum. And by conjugating the second spectrum corresponding to the first spectrum, there is no need to recalculate the second spectrum, which reduces the amount of calculation and improves the calculation efficiency.

[0214] The method provided in this embodiment can also improve the frequency response characteristics of the audio filter and reduce the amount of calculation in the audio filter design by processing the first N real part data after numerical normalization through a window function.

[0215] The method provided in this embodiment can reduce the influence of noise by performing weighted smoothing on the amplitude value difference corresponding to each frequency point, and can also make the frequency response curve smoother, reduce curve mutations, and make the frequency response curve clearer and easier to read.

[0216] Figure 6 A flowchart of an audio signal adjustment method provided by an exemplary embodiment of the present application is shown, and the method is executed by a client. Figure 3 Compared with the embodiment, the method further comprises the following steps.

[0217] Step 350: Adjust the second audio signal using the first algorithm and the second algorithm to obtain an output audio signal.

[0218] The first algorithm is used to improve the stereoscopic effect of the audio corresponding to the output audio signal, and the second algorithm is used to enhance the spatial effect of the audio corresponding to the output audio signal.

[0219] The first algorithm increases the width or depth of the sound by artificially adjusting the sound image of the output audio signal (i.e. the position of the sound in the stereo field). This is usually achieved by delaying, filtering, or adjusting the volume difference between the left and right channels. A common application of the first algorithm is to convert mono audio to stereo audio, or to enhance the spatial sense of existing stereo audio.

[0220] The second algorithm is a series of processing techniques used to improve the distribution and depth of the output audio signal in space, including the use of reverberation, delay, stereo expansion, head-related transfer function and other techniques to simulate the positioning of different sound sources in three-dimensional space. The purpose of the second algorithm is to enhance the user's sense of immersion and presence by simulating the propagation characteristics of sound in space without changing the original audio content (such as the first audio signal).

[0221] In some embodiments, the second audio signal is adjusted by a first algorithm to obtain a third audio signal; and the third audio signal is adjusted by a second algorithm to obtain an output audio signal.

[0222] First algorithm: Optionally, the first algorithm corresponds to two feasible implementation plans: Plan 1 and Plan 2.

[0223] Solution 1: In some embodiments, adjusting the second audio signal by using a first algorithm to obtain a third audio signal includes:

[0224] Acquire first left channel data and first right channel data corresponding to the second audio signal;

[0225] Calculate first edge data and first intermediate data based on the left channel data and the right channel data;

[0226] Performing weighted calculation on the first edge data and the first intermediate data to obtain second edge data and second intermediate data;

[0227] Based on the second edge data and the second intermediate data, second left channel data and second right channel data are calculated, and the third audio signal corresponds to the second left channel data and the second right channel data.

[0228] Exemplarily, the first left channel data is left, and the first right channel data is right. The first left channel data and the first right channel data can be single data or arrays including multiple data, for example: left = [10, 8, 6, 8, 20, 24], right = [12, 8, 2, 10, 16, 22].

[0229] In some embodiments, calculating the first edge data and the first intermediate data based on the left channel data and the right channel data includes:

[0230] Calculate the average of the difference between the left channel data and the right channel data to obtain the first edge data;

[0231] An average of the sum of the left channel data and the right channel data is calculated to obtain first intermediate data.

[0232] Exemplarily, the first edge data is sides, the first middle data is mid, and sides=(1 / 2)*(left-right); mid=(1 / 2)*(left+right).

[0233] In some embodiments, the weight coefficient of the weighted calculation association is greater than or equal to 0 and less than or equal to 2;

[0234] Performing weighted calculation on the first edge data and the first intermediate data to obtain second edge data and second intermediate data includes:

[0235] Multiply the first edge data and the weight coefficient to obtain the second edge data;

[0236] The first intermediate data and the first weight difference are multiplied to obtain the second intermediate data, where the first weight difference is the difference between 2 and the weight coefficient.

[0237] Exemplarily, the second edge data is newsides, the second middle data is newmid, the weight coefficient of the weighted calculation association is width, and width is greater than or equal to 0 and less than or equal to 2. newsides=width*sides, newmid=(2-width)*mid.

[0238] In some embodiments, calculating the second left channel data and the second right channel data based on the second edge data and the second intermediate data includes:

[0239] Calculate the sum of the second intermediate data and the second edge data to obtain the second left channel data;

[0240] The difference between the second intermediate data and the second edge data is calculated to obtain the second right channel data.

[0241] Exemplarily, the second left channel data is newleft, and the second right channel data is newright. newleft=newmid+newsides, and the second right channel data newright=newmid-newsides.

[0242] Solution 2: In some embodiments, adjusting the second audio signal by using the first algorithm to obtain the third audio signal includes:

[0243] Acquire first left channel data and first right channel data corresponding to the second audio signal;

[0244] One of the first left channel data or the first right channel data is delayed for a specified time length, and the other data remains unchanged, to obtain a third audio signal.

[0245] like Figure 2 As shown, the specified time length is set within 5-35 milliseconds. Assuming that the terminal device 120 is located in the center of the target car 110, the specified time length is set to 5 milliseconds; assuming that the terminal device 120 is located in the main driving position or the co-pilot position of the target car, the specified time length is set to 20 milliseconds.

[0246] Second algorithm: In some embodiments, adjusting the third audio signal by the second algorithm to obtain the output audio signal includes:

[0247] Performing filtering processing on the third audio signal through a bandpass filter to obtain a fourth audio signal, performing filtering processing on the third audio signal through a high-pass filter to obtain a fifth audio signal, and performing filtering processing on the third audio signal through a low-pass filter to obtain a sixth audio signal;

[0248] The fourth audio signal, the fifth audio signal and the sixth audio signal are superimposed to obtain an output audio signal.

[0249] A high-pass filter is a filter that allows high-frequency signals to pass through while blocking low-frequency signals below the cutoff frequency; a low-pass filter is a filter that allows low-frequency signals to pass through while blocking high-frequency signals above the cutoff frequency. A band-pass filter is a filter that allows signals within a specific frequency range to pass through while blocking frequencies outside the specific frequency range. The specific frequency range is called the passband, and the frequency range outside the passband is called the stopband. After applying the band-pass filter, a fourth audio signal containing only specific frequency components can be obtained.

[0250] Figure 7 A flowchart of an audio signal adjustment method provided by an exemplary embodiment of the present application is shown.

[0251] In some embodiments, filtering the third audio signal 710 through the bandpass filter 720 to obtain the fourth audio signal 742 includes:

[0252] Filter the third audio signal 710 through a bandpass filter 720 to obtain a filtered audio signal 721;

[0253] Perform reverberation processing on the filtered audio signal 721 through the reverberation processor 730 to obtain a reverberation audio signal 732;

[0254] The reverberation audio signal 732 is corrected by the phase correction module 740 to obtain a fourth audio signal 742 .

[0255] The reverberation processor 730 is used to add reverberation sound effects to the filtered audio signal 721. The reverberation processor 730 increases the depth and spatial sense of the sound by simulating the reflection and attenuation of the sound in the space. The reverberation processing includes early reflection (the moment when the sound first reflects from the surface) processing and late reflection (the moment when the sound continues to reflect and reverberation is formed) processing. After the filtered audio signal 721 passes through the reverberation processor 730, the obtained reverberation audio signal 732 will have an effect of sound propagating in a three-dimensional space.

[0256] The correction step of the phase correction module 740 is an optional step. Based on the phase cancellation characteristic, it is determined whether the reverberation audio signal 732 needs to be corrected, that is, whether phase inversion needs to be performed.

[0257] The phase cancellation property means that when two sound waves with the same frequency are superimposed on each other, if their phases are consistent (i.e. in phase), the sound waves will reinforce each other and the volume will increase. If the two sound waves are in completely opposite phases (i.e. anti-phase), the sound waves will cancel each other out, causing the volume to decrease or disappear completely. In the mixing process, if two audio signals contain similar frequency components, and the corresponding phases of these frequency components are opposite, cancellation may occur on these frequency components.

[0258] Phase inversion is the process of increasing the phase angle of an audio signal by 180 degrees, flipping the original waveform upside down. In audio processing, this can be achieved by polarity inversion, which is to multiply all values ​​of the audio signal by -1.

[0259] In addition, the third audio signal 710 is filtered by the high-pass filter 722 to obtain a fifth audio signal 744, and the third audio signal 710 is filtered by the low-pass filter 724 to obtain a sixth audio signal 746.

[0260] The fourth audio signal 742 , the fifth audio signal 744 , and the sixth audio signal 746 are superimposed to obtain an output audio signal 750 .

[0261] In summary, the method provided in this embodiment adjusts the second audio signal through the first algorithm and the second algorithm to obtain the output audio signal, thereby improving the stereoscopic and spatial sense of the audio corresponding to the output audio signal and reducing the problem of difficulty for users in selecting appropriate sound effects.

[0262] The method provided in this embodiment further adjusts the second audio signal through the first algorithm under different schemes, which can meet different scene requirements, and selects a suitable scheme to adjust the second audio signal according to specific scene requirements.

[0263] The method provided in this embodiment also adjusts the second left channel data and the second right channel data corresponding to the third audio signal by adjusting the edge data (the first edge data is adjusted to the second edge data) and the intermediate data (the first intermediate data is adjusted to the second intermediate data). The edge data and the intermediate data can be processed separately, and the adjustment is more flexible and simple to implement.

[0264] The method provided in this embodiment also obtains a third audio signal by delaying one of the first left channel data or the first right channel data for a specified time length, while keeping the other data unchanged. Without adding an additional audio signal, the third audio signal can be made more stereoscopic by adjusting the time difference between the two channel data.

[0265] The method provided in this embodiment also performs filtering processing on the third audio signal through different filters, and finally superimposes the audio signals to obtain an output audio signal, so that according to different scene requirements, audio signals of some frequencies can be allowed to pass through and other parts of the audio signals can be attenuated, thereby improving the spatial sense of the output audio signal.

[0266] As an example but not limitation, assume that a client is running in the terminal device, and the client corresponds to a music playing program. When a user sits in the main driving position of a car, the terminal device and the vehicle-mounted device of the car are connected via Bluetooth.

[0267] When the user intends to play a song through the in-vehicle device, he opens the client and clicks on a song in the music player program. After clicking on the song, the client plays a prompt tone, and the audio signal corresponding to the prompt tone is used as a reference audio signal.

[0268] For example, the client plays a prompt tone "The current device has established a connection with the vehicle-mounted device and can play songs". At this point, the client has completed the steps of sending the reference audio signal to the vehicle-mounted device and recording the recorded audio signal, and continues to perform the subsequent step of adjusting the audio filter. This series of steps are completed without the user's perception, and will not interrupt the user's listening experience.

[0269] Optionally, the user can also select the "music optimization function", that is, the function of adjusting the audio filter, after clicking a song in the music player program. The user can manually choose whether to use this function according to his actual needs.

[0270] After the audio filter is adjusted, the client filters the first audio signal corresponding to the song through the audio filter to obtain a second audio signal.

[0271] In order to improve the final user experience, the second audio signal can also be adjusted by the first algorithm and the second algorithm to obtain the output audio signal. The first algorithm is used to improve the stereoscopic effect of the audio corresponding to the output audio signal, and the second algorithm is used to enhance the spatial sense of the audio corresponding to the output audio signal. These steps do not require manual selection by the user, reducing the problem of difficulty for the user to select appropriate sound effects.

[0272] Finally, the audio corresponding to the output audio signal heard by the user is closer to the original playback quality of the audio corresponding to the first audio signal compared to the audio corresponding to the first audio signal played directly, thereby improving the user's listening experience in the cockpit.

[0273] In the above embodiment, steps with the same sequence number can be considered as the same step. Figure 3 The corresponding embodiments, Figure 4 The corresponding embodiments, Figure 5 The corresponding embodiments and Figure 6 The corresponding embodiments may be implemented individually or in combination, and this application does not limit this.

[0274] Figure 8 A block diagram of an audio signal adjustment device provided by an exemplary embodiment of the present application is shown. The device can be implemented as a computer device or as a part of a computer device through software or hardware or a combination of both. The device includes:

[0275] The transceiver module 810 is used to send a reference audio signal to an onboard device in a target vehicle;

[0276] The recording module 820 is used to record and obtain a recorded audio signal, where the recorded audio signal is a signal recorded by the device in the cockpit of the target vehicle after the onboard device plays the reference audio signal;

[0277] An adjustment module 830 is used to adjust the audio filter with the goal of making the recorded audio signal the same as the reference audio signal;

[0278] The filtering module 840 is used to filter the first audio signal to be played subsequently through an audio filter to obtain a second audio signal.

[0279] In a possible design of this embodiment, the adjustment module 830 is used to perform spectrum analysis and obtain a recording frequency response curve corresponding to the recorded audio signal and a reference frequency response curve corresponding to the reference audio signal respectively;

[0280] With the goal of making the recorded frequency response curve identical to the reference frequency response curve, the filter coefficients of the audio filter are calculated, and the audio filter is used to adjust the frequency response curve corresponding to the audio signal.

[0281] In a possible design of this embodiment, the adjustment module 830 is used to calculate the amplitude difference between the jth frequency point of the recorded frequency response curve and the jth frequency point of the reference frequency response curve with the goal of making the recorded frequency response curve the same as the reference frequency response curve, the frequency points of the recorded frequency response curve correspond to the frequency points of the reference frequency response curve one by one, and j is an integer less than or equal to the filter order;

[0282] Calculate the phase of each frequency point based on a preset filter order;

[0283] Based on the phase of each frequency point and the difference of the amplitude value corresponding to each frequency point, the imaginary part and the real part of the first spectrum are obtained by Euler's formula, and the second spectrum corresponding to the first spectrum is calculated by conjugation;

[0284] Performing an inverse Fourier transform on each frequency point in the first spectrum and the second spectrum to obtain a first time domain signal corresponding to both the first spectrum and the second spectrum;

[0285] Performing numerical normalization on the first N real parts of the first time domain signal, where N is the value of the filter order plus 1;

[0286] The first N real part data after numerical normalization are processed by a window function to obtain the filter coefficients of the audio filter.

[0287] In a possible design of this embodiment, the audio filter corresponds to a first group delay, and the adjustment module 830 is used to calculate the phase of each frequency point through the first group delay, and the value of the first group delay is half of the filter order.

[0288] In a possible design of this embodiment, the phase of the jth frequency point is the quotient of the first product and the length of the effective spectrum;

[0289] The first product is the product of the inverse of the first group delay, pi, and j, and the length of the effective spectrum is half the number of frequency points associated with the spectrum analysis plus 1.

[0290] In a possible design of this embodiment, the adjustment module 830 is used to divide the first N real parts in the first time domain signal by the length of the inverse Fourier transform respectively to perform numerical normalization, and the length of the inverse Fourier transform is the same as the number of frequency points associated with the spectrum analysis.

[0291] In a possible design of this embodiment, the adjustment module 830 is used to process the N real part data after numerical normalization through the Hanning window function to obtain the filter coefficients of the audio filter.

[0292] In a possible design of this embodiment, the adjustment module 830 is further used to perform weighted smoothing on the amplitude value difference corresponding to each frequency point.

[0293] In a possible design of this embodiment, the adjustment module 830 is further configured to adjust the second audio signal by using the first algorithm and the second algorithm to obtain an output audio signal;

[0294] The first algorithm is used to improve the stereoscopic effect of the audio corresponding to the output audio signal, and the second algorithm is used to enhance the spatial effect of the audio corresponding to the output audio signal.

[0295] In a possible design of this embodiment, the adjustment module 830 is used to adjust the second audio signal through a first algorithm to obtain a third audio signal; and adjust the third audio signal through a second algorithm to obtain an output audio signal.

[0296] In a possible design of this embodiment, the adjustment module 830 is used to obtain first left channel data and first right channel data corresponding to the second audio signal;

[0297] Calculate first edge data and first intermediate data based on the left channel data and the right channel data;

[0298] Performing weighted calculation on the first edge data and the first intermediate data to obtain second edge data and second intermediate data;

[0299] Based on the second edge data and the second intermediate data, second left channel data and second right channel data are calculated, and the third audio signal corresponds to the second left channel data and the second right channel data.

[0300] In a possible design of this embodiment, the adjustment module 830 is used to calculate the average of the difference between the left channel data and the right channel data to obtain the first edge data; calculate the average of the sum of the left channel data and the right channel data to obtain the first intermediate data.

[0301] In a possible design of this embodiment, the weight coefficient associated with the weighted calculation is greater than or equal to 0 and less than or equal to 2, and the adjustment module 830 is used to multiply the first edge data and the weight coefficient to obtain the second edge data; multiply the first intermediate data and the first weight difference to obtain the second intermediate data, and the first weight difference is the difference between 2 and the weight coefficient.

[0302] In a possible design of this embodiment, the adjustment module 830 is used to calculate the sum of the second intermediate data and the second edge data to obtain the second left channel data; calculate the difference between the second intermediate data and the second edge data to obtain the second right channel data.

[0303] In a possible design of this embodiment, the adjustment module 830 is used to obtain first left channel data and first right channel data corresponding to the second audio signal;

[0304] One of the first left channel data or the first right channel data is delayed for a specified time length, and the other data remains unchanged, to obtain a third audio signal.

[0305] In a possible design of this embodiment, the adjustment module 830 is configured to perform filtering processing on the third audio signal through a bandpass filter to obtain a fourth audio signal, perform filtering processing on the third audio signal through a high-pass filter to obtain a fifth audio signal, and perform filtering processing on the third audio signal through a low-pass filter to obtain a sixth audio signal;

[0306] The fourth audio signal, the fifth audio signal and the sixth audio signal are superimposed to obtain an output audio signal.

[0307] To sum up, the device provided in this embodiment obtains a recorded audio signal through recording. The recorded audio signal is a signal recorded by the client in the cabin of the target vehicle after the onboard equipment plays the reference audio signal. With the goal of making the two audio signals the same, the audio filter is adjusted so that the first audio signal played subsequently can be filtered and optimized through the audio filter to obtain a second audio signal that is more suitable for listening in the current cabin, thereby improving the listening experience of the user in the cabin.

[0308] The device provided in this embodiment also adjusts the second audio signal through the first algorithm and the second algorithm to obtain the output audio signal, thereby improving the stereoscopic and spatial sense of the audio corresponding to the output audio signal and reducing the problem of difficulty for users to select appropriate sound effects.

[0309] This embodiment is described by taking one transceiver module 810, one recording module 820, one adjustment module 830 and one filtering module 840 as an example, and the number of the transceiver modules 810, the recording module 820, the adjustment module 830 and the filtering module 840 is not limited.

[0310] For an introduction to the functions of the transceiver module 810, please refer to Figure 3 The content of step 310 in the embodiment.

[0311] For an introduction to the functions of the recording module 820, please refer to Figure 3 The content of step 320 in the embodiment.

[0312] For an introduction to the functions of the adjustment module 830, please refer to Figure 3 The content of step 330 in the embodiment, Figure 4 The contents of step 332 and step 334 in the embodiment are as follows: Figure 5 The contents of step 510 to step 570 in the embodiment, and Figure 6 The content of step 350 in the embodiment.

[0313] For an introduction to the functions of the filter module 840, please refer to Figure 3 The content of step 340 in the embodiment.

[0314] Fig. 9 The block diagram of a computer device 900 provided by an exemplary embodiment of the present application is shown. The computer device 900 may be a portable mobile terminal, such as a smart phone, a tablet computer, a Moving Picture Experts Group Audio Layer III (MP3) player, or a Moving Picture Experts Group Audio Layer IV (MP4) player. The computer device 900 may also be called a user device, a portable terminal, or other names.

[0315] Typically, the computer device 900 includes a processor 901 and a memory 902 .

[0316] The processor 901 may include one or more processing cores, such as a 4-core processor, an 8-core processor, etc. The processor 901 may be implemented in at least one hardware form of digital signal processing (DSP), field programmable gate array (FPGA), and programmable logic array (PLA). The processor 901 may also include a main processor and a coprocessor. The main processor is a processor for processing data in an awake state, also known as a central processing unit (CPU); the coprocessor is a low-power processor for processing data in a standby state. In some embodiments, the processor 901 may be integrated with a graphics processing unit (GPU), which is responsible for rendering and drawing the content to be displayed on the display screen. In some embodiments, the processor 901 may also include an artificial intelligence (AI) processor, which is used to process computing operations related to machine learning.

[0317] The memory 902 may include one or more computer-readable storage media, which may be tangible and non-transitory. The memory 902 may also include a high-speed random access memory, and a non-volatile memory, such as one or more disk storage devices, flash memory storage devices. In some embodiments, the non-transitory computer-readable storage medium in the memory 902 is used to store at least one instruction, which is used to be executed by the processor 901 to implement the audio signal adjustment method provided in the embodiment of the present application.

[0318] In some embodiments, the computer device 900 may further include: a peripheral device interface 903 and at least one peripheral device. Specifically, the peripheral device includes: at least one of a radio frequency circuit 904 , a touch display screen 905 , a camera 906 , an audio circuit 907 and a power supply 908 .

[0319] The peripheral device interface 903 may be used to connect at least one peripheral device related to input / output (I / O) to the processor 901 and the memory 902. In some embodiments, the processor 901, the memory 902, and the peripheral device interface 903 are integrated on the same chip or circuit board; in some other embodiments, any one or two of the processor 901, the memory 902, and the peripheral device interface 903 may be implemented on a separate chip or circuit board, which is not limited in this embodiment.

[0320] The radio frequency circuit 904 is used to receive and transmit radio frequency (RF) signals, also known as electromagnetic signals. The radio frequency circuit 904 communicates with communication networks and other communication devices through electromagnetic signals. The radio frequency circuit 904 converts electrical signals into electromagnetic signals for transmission, or converts received electromagnetic signals into electrical signals. Optionally, the radio frequency circuit 904 includes: an antenna system, an RF transceiver, one or more amplifiers, a tuner, an oscillator, a digital signal processor, a codec chipset, a user identity module card, etc. The radio frequency circuit 904 can communicate with other terminals through at least one wireless communication protocol. The wireless communication protocol includes but is not limited to: the World Wide Web, a metropolitan area network, an intranet, various generations of mobile communication networks (2G, 3G, 4G and 5G), a wireless local area network and / or a wireless fidelity (Wireless Fidelity, WiFi) network. In some embodiments, the radio frequency circuit 904 may also include circuits related to near field communication (NFC), which is not limited in this application.

[0321] The touch display screen 905 is used to display the user interface (UI). The UI may include graphics, text, icons, videos, and any combination thereof. The touch display screen 905 also has the ability to collect touch signals on the surface or above the surface of the touch display screen 905. The touch signal can be input to the processor 901 as a control signal for processing. The touch display screen 905 is used to provide virtual buttons and / or virtual keyboards, also known as soft buttons and / or soft keyboards. In some embodiments, the touch display screen 905 can be one, and the front panel of the computer device 900 is set; in other embodiments, the touch display screen 905 can be at least two, which are respectively set on different surfaces of the computer device 900 or are folded; in some embodiments, the touch display screen 905 can be a flexible display screen, which is set on the curved surface or folded surface of the computer device 900. Even, the touch display screen 905 can also be set to a non-rectangular irregular shape, that is, a special-shaped screen. The touch display screen 905 can be made of materials such as a liquid crystal display (LCD) and an organic light-emitting diode (OLED).

[0322] The camera 906 is used to capture images or videos. Optionally, the camera 906 includes a front camera and a rear camera. Typically, the front camera is used to realize video calls or selfies, and the rear camera is used to realize the shooting of photos or videos. In some embodiments, there are at least two rear cameras, which are any one of a main camera, a depth of field camera, and a wide-angle camera, so as to realize the fusion of the main camera and the depth of field camera to realize the background blur function, and the fusion of the main camera and the wide-angle camera to realize panoramic shooting and virtual reality (VR) shooting functions. In some embodiments, the camera 906 may also include a flash. The flash can be a monochrome temperature flash or a dual-color temperature flash. A dual-color temperature flash refers to a combination of a warm light flash and a cold light flash, which can be used for light compensation at different color temperatures.

[0323] The audio circuit 907 is used to provide an audio interface between the user and the computer device 900. The audio circuit 907 may include a microphone and a speaker. The microphone is used to collect sound waves from the user and the environment, and convert the sound waves into electrical signals and input them into the processor 901 for processing, or input them into the radio frequency circuit 904 to achieve voice communication. For the purpose of stereo acquisition or noise reduction, there may be multiple microphones, which are respectively arranged at different parts of the computer device 900. The microphone may also be an array microphone or an omnidirectional acquisition microphone. The speaker is used to convert the electrical signal from the processor 901 or the radio frequency circuit 904 into sound waves. The speaker may be a traditional film speaker or a piezoelectric ceramic speaker. When the speaker is a piezoelectric ceramic speaker, it can not only convert the electrical signal into sound waves audible to humans, but also convert the electrical signal into sound waves inaudible to humans for purposes such as distance measurement. In some embodiments, the audio circuit 907 may also include a headphone jack.

[0324] The power supply 908 is used to power various components in the computer device 900. The power supply 908 can be an alternating current, a direct current, a disposable battery, or a rechargeable battery. When the power supply 908 includes a rechargeable battery, the rechargeable battery can be a wired rechargeable battery or a wireless rechargeable battery. A wired rechargeable battery is a battery that is charged through a wired line, and a wireless rechargeable battery is a battery that is charged through a wireless coil. The rechargeable battery can also be used to support fast charging technology.

[0325] In some embodiments, the computer device 900 further includes one or more sensors 909 , including but not limited to: an acceleration sensor 910 , a gyroscope sensor 911 , a pressure sensor 912 , an optical sensor 913 , and a proximity sensor 914 .

[0326] The acceleration sensor 910 can detect the magnitude of acceleration on the three coordinate axes of the coordinate system established by the computer device 900. For example, the acceleration sensor 910 can be used to detect the components of gravity acceleration on the three coordinate axes. The processor 901 can control the touch display screen 905 to display the user interface in a horizontal view or a vertical view according to the gravity acceleration signal collected by the acceleration sensor 910. The acceleration sensor 910 can also be used for collecting game or user motion data.

[0327] The gyro sensor 911 can detect the body direction and rotation angle of the computer device 900, and the gyro sensor 911 can cooperate with the acceleration sensor 910 to collect the user's 3D actions on the computer device 900. The processor 901 can implement the following functions based on the data collected by the gyro sensor 911: motion sensing (such as changing the UI according to the user's tilt operation), image stabilization during shooting, game control, and inertial navigation.

[0328] The pressure sensor 912 can be set on the side frame of the computer device 900 and / or the lower layer of the touch display screen 905. When the pressure sensor 912 is set on the side frame of the computer device 900, it can detect the user's grip signal of the computer device 900, and perform left and right hand recognition or shortcut operation according to the grip signal. When the pressure sensor 912 is set on the lower layer of the touch display screen 905, it can control the operability controls on the UI interface according to the user's pressure operation on the touch display screen 905. The operability controls include at least one of a button control, a scroll bar control, an icon control, and a menu control.

[0329] The optical sensor 913 is used to collect the ambient light intensity. In one embodiment, the processor 901 can control the display brightness of the touch display screen 905 according to the ambient light intensity collected by the optical sensor 913. Specifically, when the ambient light intensity is high, the display brightness of the touch display screen 905 is increased; when the ambient light intensity is low, the display brightness of the touch display screen 905 is reduced. In another embodiment, the processor 901 can also dynamically adjust the shooting parameters of the camera assembly 906 according to the ambient light intensity collected by the optical sensor 913.

[0330] The proximity sensor 914, also called a distance sensor, is usually arranged on the front of the computer device 900. The proximity sensor 914 is used to collect the distance between the user and the front of the computer device 900. In one embodiment, when the proximity sensor 914 detects that the distance between the user and the front of the computer device 900 is gradually decreasing, the processor 901 controls the touch display screen 905 to switch from the screen-on state to the screen-off state; when the proximity sensor 914 detects that the distance between the user and the front of the computer device 900 is gradually increasing, the processor 901 controls the touch display screen 905 to switch from the screen-off state to the screen-on state.

[0331] Those skilled in the art will appreciate that the structure shown in this embodiment does not limit the computer device 900 , and may include more or fewer components than shown in the figure, or combine certain components, or adopt a different component arrangement.

[0332] An embodiment of the present application also provides a computer device, which includes: a processor and a memory, wherein at least one program is stored in the memory; the processor is used to execute at least one program in the memory to implement the audio signal adjustment method provided by the above-mentioned method embodiments.

[0333] An embodiment of the present application further provides a computer-readable storage medium, in which at least one program is stored. The at least one program is loaded and executed by a processor to implement the audio signal adjustment method provided by the above-mentioned method embodiments.

[0334] An embodiment of the present application also provides a computer program product, which includes computer instructions. The computer instructions are stored in a computer-readable storage medium. A processor obtains the computer instructions from the computer-readable storage medium, and the processor executes the computer instructions to implement the audio signal adjustment method provided in the above-mentioned method embodiments.

[0335] It should be understood that the "plurality" mentioned in this article refers to two or more. "And / or" describes the association relationship of the associated objects, indicating that there can be three relationships. For example, A and / or B can mean: A exists alone, A and B exist at the same time, and B exists alone. The character " / " generally indicates that the associated objects are in an "or" relationship.

[0336] Those skilled in the art should be aware that in one or more of the above examples, the functions described in the embodiments of the present application can be implemented with hardware, software, firmware, or any combination thereof. When implemented using software, these functions can be stored in a computer-readable storage medium or transmitted as one or more instructions or codes on a computer-readable storage medium. Computer-readable storage media include computer storage media and communication media, wherein the communication media include any media that facilitates the transmission of a computer program from one place to another. The storage medium can be any available medium that a general or special-purpose computer can access.

[0337] The above are only optional embodiments of the present application and are not intended to limit the present application. Any modifications, equivalent substitutions, improvements, etc. made within the spirit and principles of the present application should be included in the protection scope of the present application.

Claims

1. A method for adjusting an audio signal, characterized in that: The method is executed by a client, and includes: sending a reference audio signal to an onboard device in a target vehicle; Recording to obtain a recorded audio signal, wherein the recorded audio signal is a signal recorded by the client in the cockpit of the target vehicle after the onboard device plays the reference audio signal; Adjusting the audio filter with the goal of making the recorded audio signal identical to the reference audio signal; The first audio signal played subsequently is filtered by the audio filter to obtain a second audio signal.

2. The method according to claim 1, characterized in that The step of adjusting the audio filter with the goal of making the recorded audio signal identical to the reference audio signal comprises: Performing spectrum analysis, and respectively acquiring a recording frequency response curve corresponding to the recorded audio signal and a reference frequency response curve corresponding to the reference audio signal; With the goal of making the recorded frequency response curve identical to the reference frequency response curve, filter coefficients of an audio filter are calculated, and the audio filter is used to adjust the frequency response curve corresponding to the audio signal.

3. The method according to claim 2, characterized in that The step of calculating the filter coefficients of the audio filter with the goal of making the recorded frequency response curve the same as the reference frequency response curve comprises: With the goal of making the recorded frequency response curve identical to the reference frequency response curve, calculating the amplitude difference between the jth frequency point of the recorded frequency response curve and the jth frequency point of the reference frequency response curve, the frequency points of the recorded frequency response curve correspond to the frequency points of the reference frequency response curve one-to-one, and j is an integer less than or equal to the filter order; Calculate the phase of each frequency point based on a preset filter order; Based on the phase of each frequency point and the difference in amplitude value corresponding to each frequency point, an imaginary part and a real part of a first spectrum are obtained by using the Euler formula, and a second spectrum corresponding to the first spectrum is calculated by conjugation; Performing inverse Fourier transform on each frequency point in the first spectrum and the second spectrum to obtain a first time domain signal corresponding to both the first spectrum and the second spectrum; Performing numerical normalization on first N real parts of the first time domain signal, where N is the value of the filter order plus 1; The first N real part data after numerical normalization are processed by a window function to obtain the filter coefficient of the audio filter.

4. The method according to claim 3, characterized in that The audio filter corresponds to a first group delay, and the phase of each frequency point is calculated based on a preset filter order, including: The phase of each frequency point is calculated by using the first group delay, and the value of the first group delay is half of the filter order.

5. The method according to claim 4, characterized in that The phase of the j-th frequency point is the quotient of the first product and the length of the effective spectrum; The first product is the product of the inverse of the first group delay, pi and j, and the length of the effective spectrum is half the number of frequency points associated with the spectrum analysis plus 1.

6. The method according to any one of claims 3 to 5, characterized in that: The performing numerical normalization on the first N real parts of the first time domain signal comprises: The numerical normalization is performed by dividing the first N real parts of the first time domain signal by the length of the inverse Fourier transform, respectively, and the length of the inverse Fourier transform is the same as the number of frequency points associated with the spectrum analysis.

7. The method according to any one of claims 3 to 6, characterized in that: The step of processing the first N real part data after numerical normalization through a window function to obtain filter coefficients of an audio filter includes: The N real part data after the numerical normalization are processed by a Hanning window function to obtain the filter coefficient of the audio filter.

8. The method according to any one of claims 3 to 7, characterized in that: The method further comprises: Weighted smoothing is performed on the amplitude value differences corresponding to each frequency point.

9. The method according to any one of claims 1 to 8, characterized in that: The method further comprises: Adjust the second audio signal by using the first algorithm and the second algorithm to obtain an output audio signal; The first algorithm is used to improve the stereoscopic effect of the audio corresponding to the output audio signal, and the second algorithm is used to enhance the spatial effect of the audio corresponding to the output audio signal.

10. The method according to claim 9, characterized in that The step of adjusting the second audio signal by using the first algorithm and the second algorithm to obtain an output audio signal includes: Adjust the second audio signal by using the first algorithm to obtain a third audio signal; The third audio signal is adjusted by using the second algorithm to obtain the output audio signal.

11. The method according to claim 10, characterized in that The step of adjusting the second audio signal by using the first algorithm to obtain a third audio signal includes: Acquire first left channel data and first right channel data corresponding to the second audio signal; Based on the left channel data and the right channel data, calculating first edge data and first intermediate data; Performing weighted calculation on the first edge data and the first intermediate data to obtain second edge data and second intermediate data; Second left channel data and second right channel data are calculated based on the second edge data and the second intermediate data, and the third audio signal corresponds to the second left channel data and the second right channel data.

12. The method according to claim 11, characterized in that The step of calculating the first edge data and the first intermediate data based on the left channel data and the right channel data comprises: Calculate the average of the difference between the left channel data and the right channel data to obtain the first edge data; An average of the sum of the left channel data and the right channel data is calculated to obtain the first intermediate data.

13. The method according to claim 11 or 12, characterized in that: The weight coefficient associated with the weighted calculation is greater than or equal to 0 and less than or equal to 2; The performing weighted calculation on the first edge data and the first intermediate data to obtain second edge data and second intermediate data includes: Multiplying the first edge data and the weight coefficient to obtain the second edge data; The first intermediate data and a first weight difference are multiplied to obtain the second intermediate data, where the first weight difference is a difference between 2 and the weight coefficient.

14. The method according to any one of claims 11 to 13, characterized in that: The calculating the second left channel data and the second right channel data based on the second edge data and the second intermediate data comprises: Calculate the sum of the second intermediate data and the second edge data to obtain the second left channel data; A difference between the second intermediate data and the second edge data is calculated to obtain the second right channel data.

15. The method according to claim 10, characterized in that The step of adjusting the second audio signal by using the first algorithm to obtain a third audio signal includes: Acquire first left channel data and first right channel data corresponding to the second audio signal; One of the first left channel data or the first right channel data is delayed for a specified time period, and the other data remains unchanged, to obtain the third audio signal.

16. The method according to claim 10, characterized in that The step of adjusting the third audio signal by using the second algorithm to obtain the output audio signal includes: filtering the third audio signal through a bandpass filter to obtain a fourth audio signal, filtering the third audio signal through a high-pass filter to obtain a fifth audio signal, and filtering the third audio signal through a low-pass filter to obtain a sixth audio signal; The fourth audio signal, the fifth audio signal and the sixth audio signal are superimposed to obtain the output audio signal.

17. An audio signal adjustment device, characterized in that: The device comprises: a transceiver module, for sending a reference audio signal to an onboard device in a target vehicle; a recording module, configured to record and obtain a recorded audio signal, wherein the recorded audio signal is a signal recorded by the device in the cockpit of the target vehicle after the onboard device plays the reference audio signal; An adjustment module, configured to adjust the audio filter with the goal of making the recorded audio signal identical to the reference audio signal; The filtering module is used to filter the first audio signal played subsequently through the audio filter to obtain a second audio signal.

18. A computer device, characterized in that: The computer device comprises: a processor and a memory, wherein the memory stores at least one program; the processor is used to execute the at least one program in the memory to implement the audio signal adjustment method according to any one of claims 1 to 16.

19. A computer-readable storage medium, characterized in that: The computer-readable storage medium stores at least one program, and the at least one program is loaded and executed by the processor to implement the audio signal adjustment method according to any one of claims 1 to 16.

20. A computer program product, characterized in that The computer program product includes computer instructions, which are stored in a computer-readable storage medium. The processor obtains the computer instructions from the computer-readable storage medium, and the processor executes the computer instructions to implement the audio signal adjustment method according to any one of claims 1 to 16.

Images