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

Abstract

The application discloses an audio signal adjustment method, device, equipment, medium and program product, and belongs to the field of audio processing. The method is executed by a client, and the method comprises the following steps: sending a reference audio signal to an airborne device in a target vehicle; recording a recorded audio signal, the recorded audio signal being a signal recorded by the client in a cabin of the target vehicle after the airborne device plays the reference audio signal; adjusting an audio filter with the same target of the recorded audio signal and the reference audio signal; and filtering a first audio signal played subsequently through the audio filter to obtain a second audio signal. The method records the recorded audio signal, the recorded audio signal being an audio signal associated with the reference audio signal, adjusts the audio filter with the same target of the two audio signals, and thus can filter and optimize the first audio signal played subsequently, so that the second audio signal is more suitable for listening in the current cabin.

Description

TECHNICAL FIELD

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

[0002] With the continuous development of the automotive industry, playing music through an on-board device is a common entertainment item for users during driving.

[0003] In related technologies, the on-board device can play locally stored music, or can play music in a terminal device (such as a smartphone) after being connected to the terminal device.

[0004] However, due to the special closed environment of the vehicle cabin, which is different from the ear environment or the loudspeaker external playing environment of the terminal device, the music played by the on-board device is usually affected by the vehicle cabin, resulting in poor music quality heard by the user. SUMMARY

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

[0006] According to an aspect of an embodiment of the present application, an audio signal adjustment method is provided, which is executed by a client, and the method includes:

[0007] sending a reference audio signal to an on-board device in a target vehicle;

[0008] recording a recorded audio signal, the recorded audio signal being a signal recorded by the client in a cabin of the target vehicle after the on-board device plays the reference audio signal;

[0009] adjusting an audio filter with the same target of the recorded audio signal and the reference audio signal;

[0010] filtering a first audio signal played subsequently through the audio filter to obtain a second audio signal.

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

[0012] a transceiving module configured to send a reference audio signal to an on-board device in a target vehicle;

[0013] a recording module configured to record a recorded audio signal, the recorded audio signal being a signal recorded by the device in a cabin of the target vehicle after the on-board device plays the reference audio signal;

[0014] The adjusting module is configured to adjust the audio filter so that the recorded audio signal and the reference audio signal are the same.

[0015] The filtering module is configured to filter the first audio signal played subsequently by the audio filter to obtain a second audio signal.

[0016] According to another aspect of the embodiments of the present application, a computer device is provided, which comprises a processor and a memory, and the memory stores at least one program. The processor is configured to execute the at least one program stored in the memory to implement the audio signal adjusting method.

[0017] According to another aspect of the embodiments of the present application, a computer readable storage medium is provided, which stores at least one program. The at least one program is loaded and executed by a processor to implement the audio signal adjusting method.

[0018] According to another aspect of the embodiments of the present application, a computer program product or a computer program is provided, which comprises computer instructions stored in a computer readable storage medium. A processor acquires the computer instructions from the computer readable storage medium, and executes the computer instructions to implement the audio signal adjusting method.

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

[0020] The method records the recorded audio signal, which is a signal recorded by the client in the cabin of the target vehicle after the airborne device plays the reference audio signal. The audio filter is adjusted so that the recorded audio signal and the reference audio signal are the same. Thus, the first audio signal played subsequently can be filtered and optimized by 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. BRIEF DESCRIPTION OF DRAWINGS

[0021] In order to more clearly illustrate the technical solutions in the embodiments of the present application, the drawings needed in the embodiment description will be briefly introduced. Obviously, the drawings in the following description are only some embodiments of the present application, and other drawings can be obtained by those skilled in the art without creative effort.

[0022] Figure 1 FIG. 1 shows a schematic diagram of an audio signal adjusting system provided by an exemplary embodiment of the present application;

[0023] Figure 2 FIG. 2 shows a schematic diagram of an audio signal adjusting method provided by an exemplary embodiment of the present application;

[0024] Figure 3 A flow chart of an audio signal adjusting method is shown according to an example embodiment of the present application;

[0025] Figure 4 A flow chart of an audio signal adjusting method is shown according to an example embodiment of the present application;

[0026] Figure 5 A flow chart of an audio signal adjusting method is shown according to an example embodiment of the present application;

[0027] Figure 6 A flow chart of an audio signal adjusting method is shown according to an example embodiment of the present application;

[0028] Figure 7 A flow chart of an audio signal adjusting method is shown according to an example embodiment of the present application;

[0029] Figure 8 A block diagram of an audio signal adjusting apparatus is shown according to an example embodiment of the present application;

[0030] Figure 9 A structural block diagram of a computer device is shown according to an example embodiment of the present application. DETAILED DESCRIPTION

[0031] For the purpose of clarity, technical solutions and advantages of the present application will be further described in detail below with reference to the accompanying drawings.

[0032] The example embodiments will be described in detail below with reference to the accompanying drawings. The following description relates to the drawings, in which the same numbers denote the same or similar elements throughout the several drawings. The embodiments described in the following example embodiments do not represent all the implementations in accordance with this application. Rather, they are merely examples in accordance with some aspects of this application as detailed in the appended claims.

[0033] The terminology used in this application is for the purpose of describing particular embodiments only and is not intended to be limiting of this application. As used in this application and the appended claims, the singular forms "a," "an" and "the" are intended to include the plural forms as well, unless the context clearly indicates otherwise. It will also be understood that the term "and / or" as used herein refers to and encompasses any and all possible combinations of one or more of the 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 for analysis, stored data, displayed data, etc.) involved in the present application are all information and data authorized by the object or fully authorized by all parties, and the collection, use and processing of related 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. can be used in this application to describe various information, these information should not be limited to these terms. These terms are only used to distinguish information of the same type from each other. For example, without departing from the scope of the present application, the first parameter can also be called the second parameter, and similarly, the second parameter can also be called the first parameter. Depending on the context, the word "if" as used herein can be interpreted as "when" or "when" or "in response to determining".

[0036] First, some of the terms involved in the present application are explained:

[0037] Spectrum analysis: the process of converting a signal from time domain to 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: a graph that describes the response characteristics of a system or device to signals of different frequencies. The frequency response curve usually takes frequency as the horizontal coordinate and amplitude (or gain) as the vertical coordinate, 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. The reference audio signal is a reference signal in the audio processing process, which is used to help improve or analyze the actual acquired audio signal.

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

[0041] Euler formula: a formula in complex analysis that establishes the relationship between complex exponential function and trigonometric function, the formula is as follows:

[0042] e iθ = cos(θ) + isin(θ);

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

[0044] In signal processing, the spectrum is usually represented 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. Assume there is a complex number C with an amplitude |C| and a phase θ, which 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 the 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 unwanted components. Including: band-pass filter, high-pass filter, low-pass filter, etc.

[0052] Figure 1 A schematic diagram of an audio signal adjustment system 100 provided by an example 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 on-board 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 the like. The terminal device 120 runs a client 122, which has functions such as recording, sending reference audio signals, and controlling the terminal device 120 to play audio. For example, the client 122 is the client corresponding to the music playing program. In the embodiments of the present application, the terminal device 120 is usually taken as an example to illustrate.

[0054] In some embodiments, the target vehicle 110 includes a vehicle (traditional car, smart car, unmanned car, etc.), an airplane, a ship, a train, and the like. In the embodiments of the present application, the target vehicle 110 is usually taken as an example to illustrate.

[0055] In some embodiments, the target vehicle 110 is deployed with an on-board device 112. The on-board device 112 can be installed with a client running a target application, which can be an application with audio playing function, such as a music playing program, which is the same as the music playing program corresponding to the client 122. For example, in the case that the client 122 sends a reference audio signal to the on-board device 112, the on-board device 112 plays the audio corresponding to the reference audio signal.

[0056] With the continuous development of the automotive industry, playing music through an on-board device is a common entertainment item for users during driving. In the related art, the on-board device can play locally stored music, or can play music in a terminal device (such as a smart phone) after being connected to the terminal device. However, due to the special sealed scenario of the automobile cabin, which is different from the ear field scenario or the loudspeaker external playing scenario of the terminal device, the music played by the on-board device is usually affected by the automobile cabin, resulting in poor quality of the music heard by the user.

[0057] To solve the above problems, the present application provides an audio signal adjustment method. Figure 2 A schematic diagram of an audio signal adjustment method provided by an example embodiment of the present application is shown, which is executed by the client 122, and the terminal device 120 runs the client 122.

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

[0059] Optionally, the terminal device 120 and the on-board device 112 are connected through Bluetooth.

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

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

[0062] Optionally, the user may place the terminal device 120 in a designated location within the target vehicle 110 according to actual needs. As an example and not a limitation, when the target vehicle 110 contains only one user sitting in the driver's seat, the user may place the terminal device 120 in the driver's seat; when the target vehicle 110 contains multiple people, the user may place the terminal device 120 in the center of the target vehicle 110 or in any of the seats.

[0063] In some embodiments, the client 122 records a recorded audio signal, which is a signal recorded by the client in the cabin of the target vehicle after the onboard device 112 plays a reference audio signal.

[0064] Optionally, client 122 records an audio signal for a preset duration. The preset duration is a settable value, such as 2 seconds or 3 seconds. Too short a recording time increases the difficulty of adjusting the audio signal, thus affecting the adjustment effect; too long a recording time affects the user experience.

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

[0066] Spectrum analysis is the process of converting a 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 audio signal's frequency response curve can be obtained, showing the amplitude distribution at different frequencies.

[0067] Optionally, the client 122 performs spectrum analysis and spectrum smoothing on the recorded audio signal to obtain the 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 typically achieved through averaging, filtering, or other mathematical methods to obtain a smoother frequency response curve.

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

[0070] like Figure 2 As shown, the solid line represents the reference frequency response curve corresponding to the reference audio signal, and the dashed line represents the recording frequency response curve corresponding to the recorded audio signal. The recording frequency response curve and the reference frequency response curve are two curves that include the same number of frequency points. The j-th frequency point of the recording frequency response curve corresponds to the j-th 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 and the reference frequency response curve are different, that is, the amplitude value corresponding to the jth frequency point of the recorded frequency response curve is usually different from the amplitude value corresponding to the jth frequency point of the reference frequency response curve.

[0072] In some embodiments, the audio filter is adjusted to target the recorded audio signal and the reference audio signal being the same.

[0073] Optionally, filter coefficients of the audio filter are calculated to target the recorded frequency response curve and the reference frequency response curve being the same, the audio filter being used to adjust the frequency response curve corresponding to the audio signal.

[0074] The filter coefficients of the audio filter are a series of constants used to define the characteristics of the audio filter. The audio filter is applicable to the following formula:

[0075]

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

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

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

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

[0080] Based on the phase of each frequency point and the amplitude value difference corresponding to each frequency point, obtaining the imaginary part and the real part of a first spectrum through Euler's formula, and obtaining a second spectrum corresponding to the first spectrum through conjugate calculation;

[0081] 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 the first spectrum and the second spectrum;

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

[0083] Processing the first N real part data after numerical normalization through 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 being used to represent a degree of delay of each frequency component in the audio signal, and a phase of each frequency point is calculated by the first group delay.

[0085] The first group delay GroundDelay is calculated as follows:

[0086] GroundDelay = order / 2;

[0087] wherein order is the filter order.

[0088] The phase θ of the jth frequency point is calculated as follows:

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

[0090] wherein j is the serial number of the frequency point, PI is the circular constant, length is the length of the effective frequency spectrum, length = nfft / 2+1, nfft is the number of frequency points associated with the frequency spectrum analysis (Fourier transform), i.e. 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] wherein the ceil function is a rounding-up function, and the pow function is a power operation function, which is used here to calculate the ceil power of 2.

[0093] Suppose there is a complex number C with an amplitude |C| and a phase θ, which can be represented as:

[0094] C = |C|*e iθ ;

[0095] wherein e is the base number of natural logarithm, and i is the imaginary unit, and according to Euler's formula, C can be represented as:

[0096] C = |C|*(cos(θ) + isin(θ));

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

[0098] Re(C) = |C|*cos(θ);

[0099] Im(C) = |C|*sin(θ);

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

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

[0102] Optionally, the window function comprises at least one of a rectangular window function, a Hanning window function, a Hamming window function, and a Blackman window function. Embodiments of the present application take the Hanning window function as an example, and the window function is not limited.

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

[0104]

[0105] Wherein, N is the length of the Hanning window, and n is the serial number of the real part data.

[0106] For example, the first filter coefficient is the product of w(0) and the first real part data after numerical normalization, the second filter coefficient is the product of w(1) and the second real part data after numerical normalization, and the n-th filter coefficient is the product of w(n) and the n-th real part data after numerical normalization.

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

[0108] After the filter coefficients of the audio filter are calculated, the first audio signal played subsequently is filtered through the audio filter to adjust the first audio signal and obtain a second audio signal with better quality for the user.

[0109] In some embodiments, the method further comprises adjusting the second audio signal through a first algorithm and a second algorithm to obtain an output audio signal.

[0110] The first algorithm is used to improve the stereoscopic sense 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.

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

[0112] The first algorithm: optionally, the first algorithm corresponds to two implementable schemes: scheme one and scheme two.

[0113] Scheme one: In some embodiments, the first left channel data and the first right channel data corresponding to the second audio signal are acquired;

[0114] Based on the left channel data and the right channel data, the first side data and the first middle data are calculated;

[0115] The weighted calculation is performed on the first side data and the first middle data, and the second side data and the second middle data are acquired;

[0116] Based on the second side data and the second middle data, the second left channel data and the 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] For example, 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 a single data or an array including multiple data, for example: left = [10, 8, 6, 8, 20, 24], right = [12, 8, 2, 10, 16, 22].

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

[0119] The second side data is newsides, the second middle data is newmid, and the weight coefficient associated with the weighted calculation is width, 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] Scheme two: In some embodiments, the first left channel data and the first right channel data corresponding to the second audio signal are acquired; 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 acquire the third audio signal.

[0122] For example, the specified time length is set within 5-35 milliseconds. Suppose the terminal device 120 is in the center of the target vehicle 110, the specified time length is set to 5 milliseconds; suppose the terminal device 120 is in the main driver position or the copilot position of the target vehicle, the specified time length is set to 20 milliseconds.

[0123] The second algorithm is used to adjust the third audio signal to obtain an output audio signal, including:

[0124] The third audio signal is filtered by a band-pass filter to obtain a fourth audio signal, filtered by a high-pass filter to obtain a fifth audio signal, and filtered by 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 the output audio signal.

[0126] The high-pass filter is a filter that allows high-frequency signals to pass through while blocking low-frequency signals below the cutoff frequency; the low-pass filter is a filter that allows low-frequency signals to pass through while blocking high-frequency signals above the cutoff frequency. The 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] In summary, the method provided in this embodiment obtains a recording audio signal by recording, and the recording audio signal is a signal recorded by the client in the cabin of the target vehicle after the airborne device plays the reference audio signal. The same as the two audio signals is targeted, and the audio filter is adjusted, so that the first audio signal played subsequently can be filtered and optimized by the audio filter to obtain a second audio signal 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 further adjusts the second audio signal by the first algorithm and the second algorithm to obtain an output audio signal, thereby improving the stereoscopic and spatial sense of the audio corresponding to the output audio signal and reducing the problem of the user being difficult to select a suitable sound effect.

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

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

[0131] In some embodiments, the target vehicle includes: a vehicle (traditional car, smart car, unmanned vehicle, etc.), an airplane, a ship, a train, and the like. In the embodiments of the present application, the target vehicle is usually taken as an example of a car.

[0132] In some embodiments, the target vehicle is deployed with an on-board device, which can be a control system of the target vehicle. The on-board device can be installed with a client running a target application, which can be an application with audio playing function, such as a music playing program.

[0133] For example, the on-board device plays the audio corresponding to the reference audio signal when the client sends the reference audio signal to the on-board device.

[0134] In some embodiments, the reference audio signal is a known signal used for comparison or processing with the actually obtained audio signal, such as a recorded audio signal.

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

[0136] Step 320: recording to obtain a recorded audio signal.

[0137] The recorded audio signal is a signal recorded by the client in the cabin of the target vehicle after the on-board device plays the reference audio signal.

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

[0139] Optionally, the user places the terminal device at a designated position in the target vehicle according to actual needs. For example, when the target vehicle is a target car and only one person sitting in the driver's seat is included in the target car, the user places the terminal device at the driver's seat; when the target car includes multiple people, the user places the terminal device at the center of the target car or at the position of any seat.

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

[0141] Step 330: adjusting the audio filter with the same target of the recorded audio signal and the reference audio signal.

[0142] For example, the audio filter is adjusted by the client in the terminal device. Figure 2As shown, the solid line is used to represent the reference frequency response curve corresponding to the reference audio signal, and the dashed line is used to represent the recording frequency response curve corresponding to the recording audio signal. The recording frequency response curve and the reference frequency response curve are two curves with the same number of frequency points, the jth frequency point of the recording frequency response curve corresponds to the jth frequency point of the reference frequency response curve, and 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 pre-set value.

[0143] The recording frequency response curve and the reference frequency response curve are different, that is, the amplitude value corresponding to the jth frequency point of the recording frequency response curve is usually different from the amplitude value corresponding to the jth frequency point of the reference frequency response curve.

[0144] Since the vehicle cabin is a closed environment, the audio signal played by the airborne device will be affected by factors such as sound reflection and environmental noise, so that the audio signal heard by the user at the listening position is different from the reference audio signal, that is, the recording audio signal is different from the reference audio signal. In order to make the recording audio signal and the reference audio signal similar or even the same, it is necessary to adjust the audio filter, for example, it is necessary to adjust the filter coefficients of the audio filter.

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

[0146] The second audio signal is the audio signal filtered and optimized from the first audio signal. The second audio signal is similar to the first audio signal, and for the second audio signal, after reflection and refraction in the cabin, it reaches the position where the terminal device is located, and is closer to the original audio quality of the first audio signal. If the airborne device directly plays the first audio signal, the audio quality of the audio signal received by the user will not be close to the original audio quality of the first audio signal due to the influence of factors such as sound reflection and environmental noise.

[0147] After adjusting the audio filter, the audio filter is applied to filter the first audio signal played subsequently, so as to adjust the first audio signal and obtain a second audio signal with better quality for the user.

[0148] In summary, the method provided by the embodiment records the recording audio signal, which is the signal recorded by the client in the cabin of the target vehicle after the airborne device plays the reference audio signal. The method adjusts the audio filter to make the two audio signals the same, so that the audio filter can be used to filter and optimize the first audio signal played subsequently, and obtain a second audio signal 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 example embodiment of the present application is shown, the method is executed by a client, and the method comprises at least one of the following steps.

[0150] Step 310: sending the reference audio signal to an on-board device in the target vehicle.

[0151] Step 320: recording to obtain a recorded audio signal.

[0152] Step 332: performing spectrum analysis, and obtaining a recorded frequency response curve corresponding to the recorded audio signal and a reference frequency response curve corresponding to the reference audio signal, respectively.

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

[0154] The frequency response curve is a graph describing the response characteristics of a system or device to signals of different frequencies. The frequency response curve usually takes frequency as the horizontal coordinate and amplitude (or gain) as the vertical coordinate, and shows the amplification or attenuation of the system or device to signals of different frequencies.

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

[0156] The spectrum smoothing is a signal processing technique used to reduce noise in spectral data. It is usually realized by taking average, filtering or other mathematical methods, so as to obtain a smoother frequency response curve.

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

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

[0159] For example, the interpolation algorithm includes linear interpolation, nearest neighbor interpolation, cubic spline interpolation, cubic spline interpolation, etc., and the embodiments of the present application are not limited thereto.

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

[0161] Exemplarily, the amplitude threshold includes a maximum lifting amplitude and a maximum attenuation amplitude, the maximum lifting amplitude being a maximum gain or lifting degree when the amplitude value is adjusted, and the maximum attenuation amplitude being a maximum reduction or attenuation degree when the amplitude value is adjusted.

[0162] In some embodiments, after performing the spectrum analysis, the system low-frequency characteristic corresponding to the target vehicle is obtained according to the original frequency response curve; a low-frequency enhancement signal corresponding to the system low-frequency characteristic 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 same target of the recording frequency response curve and 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 the audio filter are a series of constants used to define the characteristics of the audio filter. The audio filter is applicable to the following formula:

[0166]

[0167] Wherein, y(n) is the output signal at time n, x(n-k) is the input signal at time n-k, k is an integer from 0 to N-1. h(k) is the filter coefficient of the audio filter, there are N in total, 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 pre-set value.

[0168] Figure 5 A flowchart of an audio signal adjustment method provided by an exemplary embodiment of the present application is shown, and in some embodiments, the filter coefficients of the audio filter are calculated with the same target of the recording frequency response curve and the reference frequency response curve, including:

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

[0170] Optionally, the way of obtaining 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. The text file is a file format that can contain data composed 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, and 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 value difference between the jth frequency point of the recorded frequency response curve and the jth frequency point of the reference frequency response curve.

[0173] The amplitude value 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, aiming at the same as the recorded frequency response curve and the reference frequency response curve. The frequency points of the recorded frequency response curve and the frequency points of the reference frequency response curve correspond 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 the preset filter order.

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

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

[0177] GroundDelay = order / 2;

[0178] Wherein, 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] Wherein, 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 of the number of frequency points associated with the spectrum analysis plus 1.

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

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

[0183] wherein j is the serial number of the frequency point, PI is the circular constant, length is the length of the effective frequency spectrum, length = nfft / 2+1, nfft is the number of frequency points associated with the frequency spectrum analysis (Fourier transform), i.e., the length of the Fourier transform, and the calculation method of nfft is as follows:

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

[0185] wherein the ceil function is a rounding-up 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 comprises performing weighted smoothing on the amplitude value difference 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 frequency spectrum are obtained through the Euler formula, and the second frequency spectrum corresponding to the first frequency spectrum is calculated through conjugation.

[0188] Suppose there is a complex number C with an amplitude |C| and a phase θ, which can be represented as:

[0189] C = |C| * e iθ ;

[0190] wherein e is the base number of natural logarithm, i is the imaginary unit, and according to the Euler formula, C can be represented as:

[0191] C = |C| * (cos(θ) + isin(θ));

[0192] Therefore, the real part Re(C) and the imaginary part Im(C) of the complex number C are represented 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 frequency spectrum can be obtained through the Euler formula.

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

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

[0198] Step 550: 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 the first spectrum and the second spectrum.

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

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

[0201] In some embodiments, the first N real parts in the first time-domain signal are divided by the length of the inverse Fourier transform respectively to perform the numerical normalization. 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 is the length of the Fourier transform (nfft). The calculation of nfft is described in step 530 and will not be repeated here.

[0203] Step 570: processing the first N real part data after numerical normalization by 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, a Hamming window, and a Blackman window.

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

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

[0207]

[0208] where N is the length of the Hanning window, and n is the serial number of the real part data.

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

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

[0211] To sum up, the method provided in the embodiment, by performing spectrum analysis and respectively acquiring a recorded frequency response curve corresponding to the recorded audio signal and a reference frequency response curve corresponding to the reference audio signal, takes the same recorded frequency response curve and the reference frequency response curve as the target, and calculates the filter coefficient of the audio filter, so as to more reasonably and accurately adjust the audio filter and improve the filtering effect of the audio filter on the first audio signal.

[0212] The method provided in the embodiment also converts the audio signal from the time domain to the frequency domain through spectrum analysis (for example, Fourier transform), which is beneficial to analyze the frequency characteristics of the audio signal and facilitates filtering, smoothing and other operations on the audio signal. The first frequency spectrum and the second frequency spectrum are converted from the frequency domain to the time domain through inverse Fourier transform, so as to reconstruct the audio signal.

[0213] The method provided in the embodiment also calculates the amplitude value 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 the preset filter order, so as to accurately calculate each frequency point corresponding to the first frequency spectrum. The second frequency spectrum corresponding to the first frequency spectrum is calculated through conjugation, without the need to recalculate the second frequency spectrum, thereby reducing the calculation amount and improving the calculation efficiency.

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

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

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

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

[0218] The first algorithm is used to improve the stereoscopic sense 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.

[0219] The first algorithm increases the width or depth of the sound by artificially adjusting the panning (i.e. the position of the sound in the stereo field) of the output audio signal. This is usually achieved by delaying, filtering or adjusting the volume difference between the left and right channels. One common application of the first algorithm is to convert mono audio to stereo audio, or to enhance the spatial impression 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 techniques such as reverb, delay, stereo expansion, head-related transfer function, etc. 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 (e.g. the first audio signal).

[0221] In some embodiments, the third audio signal is obtained by adjusting the second audio signal through the first algorithm; the output audio signal is obtained by adjusting the third audio signal through the second algorithm.

[0222] The first algorithm: optionally, the first algorithm corresponds to two implementable schemes: scheme one and scheme two.

[0223] Scheme one: in some embodiments, the third audio signal is obtained by adjusting the second audio signal through the first algorithm, comprising:

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

[0225] Based on the left channel data and the right channel data, calculating first edge data and first middle data;

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

[0227] Based on the second edge data and the second middle data, calculating second left channel data and second right channel data, the third audio signal corresponding to the second left channel data and the second right channel data.

[0228] For example, 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 a single data or an array including multiple data, for example: left = [10, 8, 6, 8, 20, 24], right = [12, 8, 2, 10, 16, 22].

[0229] In some embodiments, based on the left channel data and the right channel data, the first edge data and the first middle data are calculated, comprising:

[0230] calculating an average of a difference between the left channel data and the right channel data to obtain first side data;

[0231] calculating an average of a sum of the left channel data and the right channel data to obtain first middle data.

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

[0233] In some embodiments, the weight coefficient associated with the weighted calculation is greater than or equal to 0 and less than or equal to 2.

[0234] performing the weighted calculation on the first side data and the first middle data to obtain second side data and second middle data, comprising:

[0235] multiplying the first side data by the weight coefficient to obtain the second side data;

[0236] multiplying the first middle data by a first weight difference value to obtain the second middle data, the first weight difference value being a difference between 2 and the weight coefficient.

[0237] For example, the second side data is newsides, the second middle data is newmid, and the weight coefficient associated with the weighted calculation is width, width being greater than or equal to 0 and less than or equal to 2. newsides=width*sides, newmid=(2-width)*mid.

[0238] In some embodiments, based on the second side data and the second middle data, the second left channel data and the second right channel data are calculated, comprising:

[0239] calculating a sum of the second middle data and the second side data to obtain the second left channel data;

[0240] calculating a difference between the second middle data and the second side data to obtain the second right channel data.

[0241] For example, 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] Scheme II: In some embodiments, the second audio signal is adjusted by a first algorithm to obtain a third audio signal, comprising:

[0243] obtaining 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] As shown in Figure 2 The specified time length is set within 5-35 milliseconds, and is set to 5 milliseconds when the terminal device 120 is at the center of the target vehicle 110, and is set to 20 milliseconds when the terminal device 120 is at the main driver position or the co-driver position of the target vehicle.

[0246] The second algorithm: in some embodiments, the third audio signal is adjusted by the second algorithm to obtain an output audio signal, including:

[0247] The third audio signal is filtered by a band-pass filter to obtain a fourth audio signal, and is filtered by a high-pass filter to obtain a fifth audio signal, and is filtered by 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 the output audio signal.

[0249] The high-pass filter is a filter that allows high-frequency signals to pass through while blocking low-frequency signals below the cutoff frequency; the low-pass filter is a filter that allows low-frequency signals to pass through while blocking high-frequency signals above the cutoff frequency. The band-pass filter is a filter that allows signals in 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 example embodiment of the present application is shown.

[0251] In some embodiments, the third audio signal 710 is filtered by a band-pass filter 720 to obtain a fourth audio signal 742, including:

[0252] The third audio signal 710 is filtered by the band-pass filter 720 to obtain a filtered audio signal 721;

[0253] The filtered audio signal 721 is subjected to reverberation processing by a reverberation processor 730 to obtain a reverberation audio signal 732;

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

[0255] The reverb processor 730 is configured to add a reverberation sound effect to the filtered audio signal 721. The reverb processor 730 adds depth and spaciousness to the sound by simulating the reflection and decay of sound in a space. Reverb processing includes early reflection (the time when sound is first reflected off a surface) processing and late reflection (the time when sound continues to reflect and reverberate) processing. After the filtered audio signal 721 passes through the reverb processor 730, the obtained reverberation audio signal 732 has 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, i.e., whether the phase inversion needs to be performed.

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

[0258] The phase inversion refers to increasing the phase angle of an audio signal by 180 degrees, so that the original waveform is upside down. In audio processing, this can be achieved by polarity inversion, i.e., multiplying 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 the embodiment adjusts the second audio signal through the first algorithm and the second algorithm to obtain an output audio signal, thereby improving the stereoscopic effect and the spaciousness of the audio corresponding to the output audio signal, and reducing the problem that the user has difficulty in selecting a suitable sound effect.

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

[0263] The method provided in the embodiment can also adjust the second left channel data and the second right channel data corresponding to the third audio signal by adjusting the edge data (adjusting the first edge data to the second edge data) and the middle data (adjusting the first middle data to the second middle data), can separately process the edge data and the middle data, and has more flexibility and is simple to implement.

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

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

[0266] By way of example and not limitation, it is assumed that a client corresponding to a music playing program is running in the terminal device. When the user sits in the driver's seat of the car, the terminal device and the vehicle-mounted device of the car establish a connection through Bluetooth.

[0267] When the user intends to play a song through the vehicle-mounted device, the client is opened, and a certain song in the music playing program is clicked. After the song is clicked, the client plays a prompt tone, and the audio signal corresponding to the prompt tone is used as the reference audio signal.

[0268] For example, the client plays the prompt tone "the current device has established a connection with the vehicle-mounted device and can play a song". At this time, 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 is completed without the user's awareness and does not interrupt the user's listening experience.

[0269] Alternatively, the user can also select the "music optimization function", i.e., the function of adjusting the audio filter, after clicking a certain song in the music playing program. The user can manually select whether to use the function according to 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 the second audio signal.

[0271] To improve the final user experience, the second audio signal can also be adjusted by a first algorithm and a second algorithm to obtain an output audio signal, the first algorithm being used to improve the stereoscopic sense of the audio corresponding to the output audio signal, and the second algorithm being used to enhance the spatial sense of the audio corresponding to the output audio signal. These steps do not require the user to manually select, reducing the problem that the user has difficulty in selecting a suitable sound effect.

[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 with directly playing the audio corresponding to the first audio signal, thereby improving the user's listening experience in the cabin.

[0273] In the above embodiments, steps with the same serial numbers can be considered as the same step. Among them, Figure 3 the corresponding embodiments, Figure 4 the corresponding embodiments, Figure 5 the corresponding embodiments and Figure 6 the corresponding embodiments can be implemented alone or in combination, and the present application does not limit this.

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

[0275] The transceiver module 810 is configured to send the reference audio signal to the airborne device in the target vehicle;

[0276] The recording module 820 is configured to record a recorded audio signal, the recorded audio signal being a signal recorded by the device in the cabin of the target vehicle after the airborne device plays the reference audio signal;

[0277] The adjustment module 830 is configured to adjust the audio filter with the same recorded audio signal and reference audio signal as the target;

[0278] The filter module 840 is configured to filter the first audio signal played subsequently by the audio filter to obtain a second audio signal.

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

[0280] The adjustment module 830 is configured to calculate the filter coefficients of the audio filter with the same recorded frequency response curve and reference frequency response curve as the target, the audio filter being used to adjust the frequency response curve corresponding to the audio signal.

[0281] In a possible design of the present embodiment, the adjusting module 830 is configured to: target the same as the recorded frequency response curve and the reference frequency response curve, calculate the amplitude value 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 and the frequency points of the reference frequency response curve correspond to each other 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 the preset filter order;

[0283] based on the phase of each frequency point and the amplitude value difference corresponding to each frequency point, obtain the imaginary part and the real part of the first frequency spectrum by using the Euler formula, and obtain the second frequency spectrum corresponding to the first frequency spectrum by using conjugate calculation;

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

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

[0286] obtain the filter coefficients of the audio filter by processing the first N real part data after numerical normalization by using a window function.

[0287] In a possible design of the present embodiment, the audio filter corresponds to a first group delay, and the adjusting module 830 is configured to calculate the phase of each frequency point by using the first group delay, and the value of the first group delay is half of the filter order.

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

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

[0290] In a possible design of the present embodiment, the adjusting module 830 is configured 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 the present embodiment, the adjusting module 830 is configured to obtain the filter coefficients of the audio filter by processing the N real part data after numerical normalization by using a Hanning window function.

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

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

[0294] The first algorithm is used to improve the stereoscopic sense 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.

[0295] In a possible design of the present embodiment, the adjusting module 830 is configured to adjust the second audio signal by using a first algorithm to obtain a third audio signal, and adjust the third audio signal by using a second algorithm to obtain an output audio signal.

[0296] In a possible design of the present embodiment, the adjusting module 830 is configured to obtain first left channel data and first right channel data corresponding to the second audio signal.

[0297] The first edge data and the first middle data are calculated based on the left channel data and the right channel data.

[0298] The second edge data and the second middle data are obtained by performing weighted calculation on the first edge data and the first middle data.

[0299] The second left channel data and the second right channel data corresponding to the third audio signal are calculated based on the second edge data and the second middle data.

[0300] In a possible design of the present embodiment, the adjusting module 830 is configured to calculate the average of the difference between the left channel data and the right channel data to obtain the first edge data, and calculate the average of the sum of the left channel data and the right channel data to obtain the first middle data.

[0301] In a possible design of the present 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 adjusting module 830 is configured to multiply the first edge data by the weight coefficient to obtain the second edge data, and multiply the first middle data by a first weight difference value to obtain the second middle data, the first weight difference value being the difference between 2 and the weight coefficient.

[0302] In a possible design of the present embodiment, the adjusting module 830 is configured to calculate the sum of the second middle data and the second edge data to obtain the second left channel data, and calculate the difference between the second middle data and the second edge data to obtain the second right channel data.

[0303] In a possible design of the present embodiment, the adjusting module 830 is configured to obtain first left channel data and first right channel data corresponding to the second audio signal.

[0304] Delay one of the first left channel data or the first right channel data for a specified duration, while keeping the other data unchanged, to obtain the third audio signal.

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

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

[0307] In summary, the device provided in this embodiment obtains a recorded audio signal by recording a reference audio signal played by an onboard device. The recorded audio signal is a signal recorded by the client in the cabin of the target vehicle. With the goal of making the two audio signals the same, the audio filter is adjusted so that the subsequent playback of the first audio signal can be filtered and optimized 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 stereo and spatial sense of the audio corresponding to the output audio signal and reducing the problem of users having difficulty in selecting suitable sound effects.

[0309] This embodiment uses a transceiver module 810, a recording module 820, an adjustment module 830, and a filtering module 840 as an example for illustration. The number of transceiver modules 810, recording modules 820, adjustment modules 830, and filtering modules 840 is not limited.

[0310] For a functional description of the transceiver module 810, please refer to [link / reference]. Figure 3 The content of step 310 in the embodiment.

[0311] For a functional introduction to the recording module 820, please refer to [link / reference]. Figure 3 The content of step 320 in the embodiment.

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

[0313] For a functional description of the 840 filter module, please refer to [link / reference].Figure 3 The content of step 340 in the embodiment.

[0314] Figure 9 A structural block diagram of a computer device 900 provided by an example embodiment of the present application is shown. The computer device 900 can be a portable mobile terminal, such as a smartphone, a tablet computer, a Moving Picture Experts Group Audio Layer III (MP3) player, a Moving Picture Experts Group Audio Layer IV (MP4) player. The computer device 900 can also be referred to as a user device, a portable terminal, or other names.

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

[0316] The processor 901 can include one or more processing cores, such as a 4-core processor, an 8-core processor, and the like. The processor 901 can be implemented in at least one of a hardware form of a Digital Signal Processing (DSP), a Field Programmable Gate Array (FPGA), a Programmable Logic Array (PLA), and the like. The processor 901 can also include a main processor and a coprocessor. The main processor is a processor for processing data in an awake state, also referred to 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 can be integrated with a Graphics Processing Unit (GPU). The GPU is used to render and draw content required to be displayed on a display screen. In some embodiments, the processor 901 can further include an Artificial Intelligence (AI) processor. The AI processor is used to process computing operations related to machine learning.

[0317] The memory 902 can include one or more computer-readable storage media. The computer-readable storage media can be tangible and non-transitory; for example, magnetic discs, optical discs, or flash memory, among others. The memory 902 can also include a volatile, non-transitory memory such as a RAM, and a non-volatile, non-transitory memory such as a flash memory or a hard disk. In some embodiments, the non-transitory computer-readable storage medium in the memory 902 stores at least one instruction for execution by the processor 901 to implement the method of adjusting an audio signal provided in the embodiments of the present application.

[0318] In some embodiments, the computer device 900 can further optionally 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 can be used to connect the 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 can be implemented on a separate chip or circuit board, and the embodiments are not limited in this regard.

[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 a communication network 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 subscriber identity module card, and the like. 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 (WiFi) network. In some embodiments, the radio frequency circuit 904 can also include near field communication (NFC) related circuitry, and the present application is not limited in this regard.

[0321] The touch display screen 905 is configured to display a user interface (UI). The UI can include graphics, text, icons, video, and any combination thereof. The touch display screen 905 is further configured to capture touch signals on or above the surface of the touch display screen 905. The touch signals can be input to the processor 901 as control signals. The touch display screen 905 is configured to provide virtual buttons and / or virtual keyboard, also known as soft buttons and / or soft keyboard. In some embodiments, the touch display screen 905 can be one, configured to be the front panel of the computer device 900; in some other embodiments, the touch display screen 905 can be at least two, configured to be on different surfaces of the computer device 900 or in a folding design; in some embodiments, the touch display screen 905 can be a flexible display screen, configured to be on a curved surface or a folding surface of the computer device 900. Even, the touch display screen 905 can be configured to be an irregular shape other than a rectangle, i.e., a notched screen. The touch display screen 905 can be made of a material such as a liquid crystal display (LCD), an organic light-emitting diode (OLED), etc.

[0322] The camera 906 is configured to capture images or videos. Optionally, the camera 906 includes a front camera and a rear camera. Generally, the front camera is configured to implement video call or selfie, and the rear camera is configured to implement photo or video shooting. In some embodiments, the rear camera is at least two, configured to be any one of a main camera, a depth-of-field camera, and a wide-angle camera, to implement background blurring function by fusing the main camera and the depth-of-field camera, and to implement panorama shooting and virtual reality (VR) shooting function by fusing the main camera and the wide-angle camera. In some embodiments, the camera 906 can further include a flash. The flash can be a single-color temperature flash or a dual-color temperature flash. The 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 under different color temperatures.

[0323] The audio circuit 907 is configured to provide an audio interface between a user and the computer device 900. The audio circuit 907 can include a microphone and a speaker. The microphone is configured to collect sound waves from the user and the environment, and convert the sound waves into an electrical signal input to the processor 901 for processing, or to the radio frequency circuit 904 for voice communication. The microphone can be multiple microphones arranged at different positions of the computer device 900 for stereo sound collection or noise reduction. The microphone can also be an array microphone or an omnidirectional microphone. The speaker is configured to convert an electrical signal from the processor 901 or the radio frequency circuit 904 into sound waves. The speaker can be a traditional thin-film speaker or a piezoelectric ceramic speaker. When the speaker is a piezoelectric ceramic speaker, it can convert electrical signals into sound waves audible to humans, or sound waves inaudible to humans for ranging purposes. In some embodiments, the audio circuit 907 can further include a headphone jack.

[0324] The power supply 908 is configured to supply power to various components of 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. The wired rechargeable battery is a battery that is charged through a wired line, and the 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. The one or more sensors 909 include, but are 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 acceleration in three coordinate axes of a coordinate system established by the computer device 900. For example, the acceleration sensor 910 can be configured to detect the components of the gravitational acceleration in three coordinate axes. The processor 901 can control the touch display 905 to display a user interface in a landscape view or a portrait view according to the gravitational acceleration signals collected by the acceleration sensor 910. The acceleration sensor 910 can also be used for game or user motion data collection.

[0327] The gyroscope sensor 911 can detect the orientation and rotation angle of the computer device 900. The gyroscope sensor 911 can work with the acceleration sensor 910 to collect 3D motions of the user with respect to the computer device 900. The processor 901 can implement the following functions according to the data collected by the gyroscope sensor 911: motion sensing (e.g., changing a UI according to a user's tilt operation), image stabilization during shooting, game control, and inertial navigation.

[0328] The pressure sensor 912 can be disposed on the side bezel of the computer device 900 and / or on the lower layer of the touch display screen 905. When the pressure sensor 912 is disposed on the side bezel of the computer device 900, it can detect the user's grip signal on the computer device 900 and perform left / right hand recognition or quick operation based on the grip signal. When the pressure sensor 912 is disposed on the lower layer of the touch display screen 905, it can control operable controls on the UI interface based on the user's pressure operation on the touch display screen 905. Operable controls include at least one of button controls, scroll bar controls, icon controls, and menu controls.

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

[0330] A proximity sensor 914, also known as a distance sensor, is typically located on the front of the computer device 900. The proximity sensor 914 is used to detect 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 touchscreen display 905 to switch from a screen-on state to a 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 touchscreen display 905 to switch from a screen-off state to a screen-on state.

[0331] Those skilled in the art will understand that the structure shown in this embodiment does not constitute a limitation on the computer device 900, and may include more or fewer components than shown, or combine certain components, or use different component arrangements.

[0332] This application also provides a computer device, which includes 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 provided in the above method embodiments.

[0333] This application also provides a computer-readable storage medium storing at least one program, which is loaded and executed by a processor to implement the audio signal adjustment method provided in the above-described method embodiments.

[0334] The embodiment of the present application further provides a computer program product, which comprises computer instructions stored in a computer readable storage medium, a processor acquires the computer instructions from the computer readable storage medium, and the processor executes the computer instructions to realize the audio signal adjustment method provided by the above-mentioned method embodiments.

[0335] It should be understood that "multiple" mentioned herein refers to two or more than two. The "and / or" describes the association relationship of the associated objects, which means that there can be three relationships, for example, A and / or B can represent the following three cases: A exists alone, A and B exist simultaneously, and B exists alone. The character " / " generally represents that the associated objects before and after it are in an "or" relationship.

[0336] Those skilled in the art should realize that, in one or more examples described above, the functions described in the embodiments of the present application can be realized by hardware, software, firmware or any combination thereof. When realized by 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. The computer readable storage medium includes a computer storage medium and a communication medium, wherein the communication medium includes any medium facilitating the transmission of computer programs from one place to another. The storage medium can be any available medium accessible by a general or special purpose computer.

[0337] The above is only an optional embodiment of the present application, and does not limit the present application. Any modification, equivalent replacement, improvement, etc. made within the spirit and principle of the present application shall be included in the protection scope of the present application.

Claims

1. An audio signal adjustment method, characterized by, The method is executed by a client, and the method comprises: sending a reference audio signal to an on-board device in a target vehicle; recording a recorded audio signal, the recorded audio signal being a signal recorded by the client in a cabin of the target vehicle after the on-board device plays the reference audio signal; performing a spectrum analysis, and obtaining a recorded frequency response curve corresponding to the recorded audio signal and a reference frequency response curve corresponding to the reference audio signal respectively; targeting at the recorded frequency response curve being the same as the reference frequency response curve, calculating an amplitude value difference between a jth frequency point of the recorded frequency response curve and a jth frequency point of the reference frequency response curve, the frequency points of the recorded frequency response curve and the frequency points of the reference frequency response curve corresponding to each other one by one, and j being an integer less than or equal to a filter order; calculating a phase of each frequency point based on a preset filter order; based on the phase of each frequency point and the amplitude value difference corresponding to each frequency point, obtaining an imaginary part and a real part of a first spectrum through Euler's formula, and obtaining a second spectrum corresponding to the first spectrum through conjugate calculation; 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 the first spectrum and the second spectrum; performing numerical normalization on the first N real parts in the first time domain signal, N being a value of the filter order plus 1; processing the first N real part data after numerical normalization through a window function to obtain filter coefficients of an audio filter, the audio filter being used to adjust a frequency response curve corresponding to an audio signal; filtering a first audio signal played subsequently through the audio filter to obtain a second audio signal.

2. The method of claim 1, wherein, The audio filter corresponds to a first group delay, and the calculation of the phase of each frequency point based on the preset filter order comprises: calculating the phase of each frequency point through the first group delay, a value of the first group delay being half of the filter order.

3. The method of claim 2, wherein, The phase of the jth frequency point is a quotient of a first product and a length of an effective spectrum; wherein the first product is a product of a reciprocal of the first group delay, a constant pi and j, and the length of the effective spectrum is half of a number of frequency points associated with the spectrum analysis plus 1.

4. The method according to any one of claims 1 to 3, characterized in that, The numerical normalization on the first N real parts in the first time domain signal comprises: dividing the first N real parts in the first time domain signal by a length of inverse Fourier transform respectively to perform the numerical normalization, the length of the inverse Fourier transform being the same as the number of frequency points associated with the spectrum analysis.

5. The method according to any one of claims 1 to 3, characterized in that, The processing of the first N real part data after numerical normalization through a window function to obtain filter coefficients of an audio filter comprises: processing the first N real part data after numerical normalization through a Hanning window function to obtain the filter coefficients of the audio filter.

6. The method according to any one of claims 1 to 3, characterized in that, The method further comprises: performing weighted smoothing on the amplitude value difference corresponding to each frequency point.

7. The method according to any one of claims 1 to 3, characterized in that, The method further comprises: adjusting the second audio signal through a first algorithm and a second algorithm to obtain an output audio signal; The first algorithm is used to improve the stereoscopic sense 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.

8. The method of claim 7, wherein, The adjusting the second audio signal by the first algorithm and the second algorithm to obtain an output audio signal comprises: adjusting the second audio signal by the first algorithm to obtain a third audio signal; adjusting the third audio signal by the second algorithm to obtain the output audio signal.

9. The method of claim 8, wherein, The adjusting the second audio signal by the first algorithm to obtain a third audio signal comprises: obtaining first left channel data and first right channel data corresponding to the second audio signal; calculating first edge data and first middle data based on the left channel data and the right channel data; performing weighted calculation on the first edge data and the first middle data to obtain second edge data and second middle data; calculating second left channel data and second right channel data based on the second edge data and the second middle data, the third audio signal corresponding to the second left channel data and the second right channel data.

10. The method of claim 9, wherein, The calculating first edge data and first middle data based on the left channel data and the right channel data comprises: calculating the average of the difference between the left channel data and the right channel data to obtain the first edge data; calculating the average of the sum of the left channel data and the right channel data to obtain the first middle data.

11. The method of claim 9, wherein, 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 middle data to obtain second edge data and second middle data comprises: multiplying the first edge data and the weight coefficient to obtain the second edge data; multiplying the first middle data and a first weight difference value to obtain the second middle data, the first weight difference value being the difference between 2 and the weight coefficient.

12. The method of claim 9, wherein, The calculating second left channel data and second right channel data based on the second edge data and the second middle data comprises: calculating the sum of the second middle data and the second edge data to obtain the second left channel data; calculating the difference between the second middle data and the second edge data to obtain the second right channel data.

13. The method of claim 8, wherein, The adjusting the second audio signal by the first algorithm to obtain a third audio signal comprises: obtaining first left channel data and first right channel data corresponding to the second audio signal; delaying one of the first left channel data or the first right channel data by a specified time length, and keeping the other data unchanged to obtain the third audio signal.

14. The method of claim 8, wherein, The adjusting the third audio signal by the second algorithm to obtain the output audio signal comprises: filtering the third audio signal by a band-pass filter to obtain a fourth audio signal, filtering the third audio signal by a high-pass filter to obtain a fifth audio signal, and filtering the third audio signal by a low-pass filter to obtain a sixth audio signal; Superimpose the fourth audio signal, the fifth audio signal and the sixth audio signal to obtain the output audio signal.

15. An audio signal adjusting apparatus characterized by comprising: The device comprises: a transceiving module configured to send a reference audio signal to an on-board device in a target vehicle; a recording module configured to record a recorded audio signal, the recorded audio signal being a signal recorded by the device in a cabin of the target vehicle after the on-board device plays the reference audio signal; an adjusting module configured to perform a spectrum analysis, and obtain a recorded frequency response curve corresponding to the recorded audio signal and a reference frequency response curve corresponding to the reference audio signal; take the recorded frequency response curve being the same as the reference frequency response curve as a target, calculate an amplitude value difference between a jth frequency point of the recorded frequency response curve and a jth frequency point of the reference frequency response curve, the frequency points of the recorded frequency response curve and the frequency points of the reference frequency response curve corresponding to each other one by one, j being an integer less than or equal to a filter order; calculate a phase of each frequency point based on a preset filter order; obtain an imaginary part and a real part of a first spectrum based on the phase of each frequency point and an amplitude value difference corresponding to each frequency point through Euler's formula, and obtain a second spectrum corresponding to the first spectrum through conjugate calculation; 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 the first spectrum and the second spectrum; perform numerical normalization on the first N real parts in the first time domain signal, N being a value of the filter order plus 1; process the first N real part data after numerical normalization through a window function to obtain filter coefficients of an audio filter, the audio filter being configured to adjust a frequency response curve corresponding to an audio signal; a filtering module configured to filter a first audio signal played subsequently through the audio filter to obtain a second audio signal.

16. A computer device, comprising: The computer device comprises a processor and a memory, the memory storing at least one program; the processor is configured to execute the at least one program in the memory to implement the audio signal adjusting method according to any one of claims 1 to 14.

17. A computer-readable storage medium, characterized in that, The computer readable storage medium stores at least one program, the at least one program is loaded and executed by a processor to implement the audio signal adjusting method according to any one of claims 1 to 14.

18. A computer program product, characterised in that, The computer program product comprises computer instructions 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 adjusting method according to any one of claims 1 to 14.

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