Audio equalization method and apparatus, device, and storage medium

By dividing the audio signal into sub-bands using filter banks and calculating the delay time for processing, the overflow and distortion problems caused by audio signal adjustment in existing technologies are solved, achieving efficient audio equalization adjustment, simplifying hardware modifications, and improving audio quality.

WO2026061485A1PCT designated stage Publication Date: 2026-03-26GUANGZHOU KUGOU COMP TECH CO LTD
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Patent Information

Authority / Receiving Office
WO · WO
Patent Type
Applications
Current Assignee / Owner
Filing Date
2025-09-19
Publication Date
2026-03-26

AI Technical Summary

Technical Problem

Existing audio equalization methods are prone to audio signal overflow or clipping distortion when adjusting frequencies, and require modifications to hardware circuits to compensate for energy deficiencies, resulting in high costs.

Method used

The audio signal is filtered by a filter bank, divided into multiple audio sub-bands, and the delay time of each sub-band is calculated based on the gain of the frequency point. After delay processing, the output signal is synthesized, avoiding direct adjustment of the amplitude of the audio signal.

Benefits of technology

It achieves equalization adjustment without changing the audio signal energy, avoiding signal overflow and distortion, improving audio quality, reducing the need for hardware circuit modifications, and improving efficiency.

✦ Generated by Eureka AI based on patent content.

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Abstract

An audio equalization method and apparatus, a device, and a storage medium, relating to the technical field of audio. The method comprises: acquiring an input audio signal and equalization adjustment parameters (210); by means of a filter bank, filtering the input audio signal to obtain a plurality of audio sub-bands (220); on the basis of at least one frequency point and a gain respectively corresponding to the at least one frequency point, obtaining delay times respectively corresponding to the plurality of audio sub-bands (230); on the basis of the delay times respectively corresponding to the plurality of audio sub-bands, delaying the plurality of audio sub-bands to obtain a plurality of delayed audio sub-bands (240); and synthesizing the plurality of delayed audio sub-bands to obtain an output audio signal (250). The present application avoids affecting the energy magnitude of audio signals, implements equalization adjustment of the audio signals while ensuring the normal output of the audio signals, and improves the quality of audio equalization.
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Description

Audio equalization method, device, equipment and storage medium

[0001] The present application claims priority to the Chinese patent application No. 202411318577.7, filed on September 20, 2024, and entitled "Audio equalization method, device, equipment and storage medium", the entire content of which is incorporated herein by reference. TECHNICAL FIELD

[0002] The present application relates to the field of audio technology, in particular to an audio equalization method, device, equipment and storage medium. BACKGROUND

[0003] Equalization adjustment (audio equalizer) of an audio signal is usually implemented using an IIR (Infinite Impulse Response) or FIR (Finite Impulse Response) filter, by controlling the amplification of some frequency points in the audio signal to achieve amplification or attenuation of the amplitude of the audio signal at the corresponding frequency.

[0004] In related technologies, in a music playing program, after an audio decoder decodes an audio file into PCM (Pulse Code Modulation) data, equalization adjustment needs to be performed, and then the PCM data after equalization adjustment is sent to a power amplifier module through an audio output interface of an operating system. The energy of a digital audio source usually has a standard, such as CD (Compact Disc) standard, MP3 (Moving Picture Experts Group Audio Layer III) with a label, Dolby, AAC (Advanced Audio Coding), etc., most of which are close to the full amplitude of the audio digital signal. Therefore, if a user wants to apply an equalization adjustment with a gain of 10 dB at 120 Hz, there is a high probability that the audio digital signal will overflow, resulting in an audio signal compression effect (such as using dynamic range compression to control signal overflow after equalization adjustment) or clipping distortion. An effective solution is to first attenuate the volume of the audio digital signal (reduce the audio energy) before equalization adjustment, and then perform equalization adjustment, and finally the power amplifier module needs to be coordinated (increase the gain of the audio analog signal, or even increase the power supply voltage of the audio analog signal amplification circuit) to compensate for the attenuation of the volume of the audio digital signal.

[0005] However, the above-mentioned audio equalization method may need to modify the hardware circuit to compensate for the defect of the audio energy, which is too costly. SUMMARY

[0006] Embodiments of the present application provide an audio equalization method, device, equipment and storage medium. The technical solutions provided by the embodiments of the present application are as follows:

[0007] According to an aspect of the embodiments of the present application, an audio equalization method is provided, the method is executed by a computer device, and the method comprises the following steps:

[0008] obtaining an input audio signal and equalization adjustment parameters, wherein the equalization adjustment parameters comprise at least one frequency point that needs to be adjusted and a gain corresponding to each of the at least one frequency point, and the gain is used to amplify the frequency of the frequency point;

[0009] filtering the input audio signal through a filter bank to obtain a plurality of audio subbands, wherein the filter bank comprises a low-pass filter, a high-pass filter and at least one band-pass filter, and the at least one band-pass filter is a band-pass filter divided based on a critical band of a Bark scale;

[0010] obtaining a delay time corresponding to each of the plurality of audio subbands according to the at least one frequency point and the gain corresponding to each of the at least one frequency point, wherein the delay time is a time for unifying the auditory perception of the audio subbands;

[0011] delay processing the plurality of audio subbands according to the delay time corresponding to each of the plurality of audio subbands to obtain a plurality of delayed audio subbands;

[0012] combining the plurality of delayed audio subbands to obtain an output audio signal.

[0013] According to an aspect of the embodiments of the present application, an audio equalization device is provided, and the device comprises:

[0014] a signal obtaining module, configured to obtain an input audio signal and equalization adjustment parameters, wherein the equalization adjustment parameters comprise at least one frequency point that needs to be adjusted and a gain corresponding to each of the at least one frequency point, and the gain is used to amplify the frequency of the frequency point;

[0015] a signal filtering module, configured to filter the input audio signal through a filter bank to obtain a plurality of audio subbands, wherein the filter bank comprises a low-pass filter, a high-pass filter and at least one band-pass filter, and the at least one band-pass filter is a band-pass filter divided based on a critical band of a Bark scale;

[0016] a delay determining module, configured to obtain a delay time corresponding to each of the plurality of audio subbands according to the at least one frequency point and the gain corresponding to each of the at least one frequency point, wherein the delay time is a time for unifying the auditory perception of the audio subbands;

[0017] delay processing module, configured to perform delay processing on the plurality of audio subbands according to delay times corresponding to the plurality of audio subbands respectively, to obtain a plurality of delayed audio subbands;

[0018] subband synthesis module, configured to synthesize the plurality of delayed audio subbands to obtain an output audio signal.

[0019] According to an aspect of an embodiment of the present application, a computer device is provided, which comprises a processor and a memory, and the memory stores a computer program, which is loaded and executed by the processor to implement the above-mentioned audio equalization method.

[0020] According to an aspect of an embodiment of the present application, a computer readable storage medium is provided, which stores a computer program, and the computer program is loaded and executed by a processor to implement the above-mentioned audio equalization method.

[0021] According to an aspect of an embodiment of the present application, a computer program product is provided, which comprises a computer program, and the computer program is loaded and executed by a processor to implement the above-mentioned audio equalization method.

[0022] The technical scheme provided by the embodiments of the present application can bring the following beneficial effects:

[0023] The plurality of audio subbands are obtained by filtering the input audio signal, so that the gain of the frequency point in the equalization adjustment process of the audio signal can be converted into the delay time corresponding to the audio subband, avoiding directly amplifying or attenuating the amplitude of the audio signal, thereby avoiding affecting the energy size of the audio signal and causing the audio signal to overflow and appear compression effect or clipping distortion. The equalization adjustment of the audio signal is realized while ensuring the normal output of the audio signal. The delay time corresponding to the audio subband is calculated and the delay processing is performed on each audio subband, so that the delayed audio subband unifies the time of the auditory perception of the audio subband, making the auditory perception of the output audio signal consistent at each frequency, and improving the quality of the audio equalization. In addition, the technical method provided by the present application is easy to implement in operation, avoids modifying the hardware circuit, reduces the implementation steps of the audio equalization, saves the resources required for the audio equalization, improves the efficiency of the audio equalization, and avoids the cost being too large to make up for the defects of the audio energy. BRIEF DESCRIPTION OF DRAWINGS

[0024] FIG. 1 is a schematic diagram of a scheme implementation environment provided by an embodiment of the present application;

[0025] FIG. 2 is a flowchart of an audio equalization method provided by an embodiment of the present application;

[0026] FIG. 3 is a flow diagram of an audio equalization method according to an embodiment of the present application;

[0027] FIG. 4 is a diagram of a first frequency point assigned to a gain value of each audio sub-band according to an embodiment of the present application;

[0028] FIG. 5 is a block diagram of an audio equalization device according to an embodiment of the present application;

[0029] FIG. 6 is a structural block diagram of a computer device according to an embodiment of the present application. DETAILED DESCRIPTION

[0030] For the purpose of making the purpose, technical scheme and advantages of the present application more clear, the embodiments of the present application will be described in further detail below with reference to the drawings.

[0031] Please refer to FIG. 1, which shows a diagram of a scheme implementation environment according to an embodiment of the present application. The scheme implementation environment can be implemented as an audio equalization system. The scheme implementation environment can include a terminal device 10 and a server 20.

[0032] The number of terminal devices 10 can be one or more. The terminal device 10 can be an electronic device such as a mobile phone, a tablet computer, a notebook computer, a desktop computer, a game console, an e-book reader, a multimedia playback device, a wearable device, a smart voice interaction device, a smart home appliance, a vehicle-mounted terminal, an aircraft, etc.

[0033] The terminal device 10 can be installed with a client of a target application program, which has a function of equalizing and adjusting an input audio signal. A user can input an audio signal and an equalization adjustment parameter in the target application program, and based on at least one frequency point to be adjusted and a gain corresponding to each of the at least one frequency point, the equalization and adjustment of the audio signal can be realized. The type of the target application program is not limited in the present application, including but not limited to a music application program, a video application program, a webpage application program, etc. Optionally, the target application program can be an application program that needs to be downloaded and installed, or an application program that can be used immediately after being clicked, which is not limited in the present application.

[0034] The server 20 is configured to provide a background service for the client of the target application program installed and run in the terminal device 10. For example, the server 20 can be a background server of the above-mentioned application program. The server 20 can be a physical server, or a server cluster composed of multiple servers, or a cloud computing service center. Optionally, the server 20 can provide a background service for multiple application programs in multiple terminal devices 10. The terminal device 10 and the server 20 can communicate with each other through a network.

[0035] In the embodiment of the present application, the user inputs the audio signal to be equalized and the equalization adjustment parameter in the target application, the computer device acquires the input audio signal, and at least one frequency point to be adjusted and the gain corresponding to each of the at least one frequency point. The input audio signal is filtered through a plurality of filters in the filter bank to obtain a plurality of audio subbands, the delay time corresponding to each of the plurality of audio subbands is calculated according to the at least one frequency point and the gain corresponding to each of the at least one frequency point, and the plurality of audio subbands are respectively delayed based on the delay time. The output audio signal is obtained by synthesizing the delayed plurality of audio subbands.

[0036] Please refer to FIG. 2, which shows a flowchart of the audio equalization method provided by an embodiment of the present application. The execution subject of each step of the method can be a computer device. The method can include at least one of the following steps 210-250:

[0037] Step 210, acquiring an input audio signal and an equalization adjustment parameter, the equalization adjustment parameter including at least one frequency point to be adjusted and the gain corresponding to each of the at least one frequency point, the gain being used to amplify the frequency of the frequency point.

[0038] The input audio signal refers to the input audio digital signal, which is the PCM data obtained by decoding the audio file through the audio decoder.

[0039] The equalization adjustment parameter is the parameter input by the user for equalizing the input audio signal, and the equalization adjustment parameter includes at least one frequency point to be adjusted and the gain corresponding to each of the at least one frequency point. The frequency point is used to indicate the position of the signal amplitude to be adjusted in the input audio signal, and the gain corresponding to the frequency point is used to control the amplitude of the frequency point to amplify or attenuate the amplitude of the audio signal of the frequency point. For example, if the gain corresponding to the frequency point is 10 dB, it means that the amplitude of the audio signal of the frequency point is amplified by 3.16 times. If the gain corresponding to the frequency point is -10 dB, it means that the amplitude of the audio signal of the frequency point is amplified by 0.316 times.

[0040] Step 220, filtering the input audio signal through a filter bank to obtain a plurality of audio subbands, the filter bank including a low-pass filter, a high-pass filter, and at least one band-pass filter, the at least one band-pass filter being a band-pass filter divided based on the critical frequency band of the Bark scale.

[0041] The low-pass filter is used to acquire the audio signal lower than the cutoff frequency of the low-pass filter in the input audio signal, the high-pass filter is used to acquire the audio signal higher than the cutoff frequency of the high-pass filter in the input audio signal, and the band-pass filter is used to acquire the audio signal within the frequency band range in the input audio signal.

[0042] The filter bank can be a high stop-band attenuation IIR filter bank or a high stop-band attenuation FIR filter bank, and the application does not make any limitation. Exemplarily, the filter bank can obtain a FIR low-pass filter, a FIR high-pass filter and 23 FIR band-pass filters by using a window function design, the window function uses a Blackman window, and the filter order uses 513 orders.

[0043] In some embodiments, the performance of each filter included in the filter bank satisfies that the stop-band attenuation exceeds 60 dB and the slope exceeds 36 dB / oct.

[0044] In some embodiments, at least one band-pass filter is a band-pass filter divided based on the critical bands of the Bark scale.

[0045] Exemplarily, the filter bank includes 25 filters, the low-pass filter corresponds to a cut-off frequency of 100 Hz, the high-pass filter corresponds to a cut-off frequency of 15500 Hz, and the low cut-off frequency and the high cut-off frequency of the remaining 23 band-pass filters correspond to the corresponding start frequency and cut-off frequency of the 2nd to 24th corresponding start frequency and cut-off frequency of the critical bands of the Bark scale, respectively, that is, the frequency band range of the 1st band-pass filter is 100-200 Hz, the frequency band range of the 2nd band-pass filter is 200-300 Hz, and the frequency band range of the 23rd band-pass filter is 12000-15500 Hz.

[0046] Exemplarily, if the filter bank includes 25 filters, the input audio signal is filtered by the filter bank to obtain 25 audio subbands.

[0047] Since the signal of the audio subband is delayed and then synthesized into PCM, a large phase distortion will occur at the boundary frequency of each audio subband. If the frequency division point of the filter is selected at will, the distortion may occur in the frequency interval that is most obvious to human auditory perception. When the Bark scale is selected as the frequency division point, the phase distortion is concentrated on the boundary of each Bark frequency segment, and the Bark scale is divided according to human auditory perception. Therefore, the problem of phase distortion can be effectively solved.

[0048] In step 230, according to at least one frequency point and at least one gain corresponding to each frequency point, a plurality of audio subbands are obtained respectively corresponding to a delay time, and the delay time is a time for unifying the auditory perception of the audio subband.

[0049] According to psychoacoustics, for a complex audio signal composed of multiple frequencies, the frequency signal heard first is more obvious to auditory perception, and therefore, the delay time corresponding to each audio subband needs to be determined according to at least one frequency point and the gain corresponding to the at least one frequency point, the gain is converted into delay time without changing the energy of the audio signal, and the multiple audio subbands are aligned through the delay time, so as to realize the equalization adjustment of the audio signal.

[0050] The specific determination process of the delay time corresponding to the audio subband can refer to the following embodiments, which will not be introduced here.

[0051] In step 240, the multiple audio subbands are delayed according to the delay time corresponding to each audio subband, and the multiple audio subbands after delay are obtained.

[0052] The multiple audio subbands can be delayed by the delay device according to the delay time corresponding to each audio subband, and the multiple audio subbands after delay are obtained. For example, the number of audio samples corresponding to the delay time can be determined according to the interval time of the delay time, so that the audio sample values in the delay time are assigned to 0, and the audio subband after delay is obtained.

[0053] Exemplarily, the audio sample values of the audio subband at each position are represented as (1, 1, 0, 1, 1, 1), and the delay time contains two interval times of audio samples, and then the audio sample values of the audio subband after delay at each position are represented as (0, 0, 1, 1, 0, 1, 1, 1).

[0054] In step 250, the multiple audio subbands after delay are synthesized, and the output audio signal is obtained.

[0055] In some embodiments, step 250 includes at least one of sub-steps 251-252.

[0056] In sub-step 251, the audio sample values at the same position in the multiple audio subbands after delay are added, and the audio sub-signals corresponding to the multiple positions are obtained.

[0057] For example, if the filter bank contains 25 filters, the filtering obtains 25 audio subbands, and the 25 delayed audio subbands are S1, S2, S3, …, S24, S25 respectively, each of which contains at least one audio sample value, S1[0] represents the first audio sample value in the audio subband S1, S1[1] represents the second audio sample value in the audio subband S1, S2[0] represents the first audio sample value in the audio subband S2, and so on. For calculating the audio sub-signal corresponding to the first position, the audio sample values at the first position in the 25 delayed audio subbands are added respectively to obtain the audio sub-signal PCM[0] corresponding to the first position, i.e. PCM[0] = S1[0] + S2[0] + S3[0] + … + S25[0]. Similarly, the audio sub-signal PCM[1] corresponding to the second position is PCM[1] = S1[1] + S2[1] + S3[1] + … + S25[1], and the audio sub-signal PCM[n] corresponding to the nth position is PCM[n] = S1[n] + S2[n] + S3[n] + … + S25[n], where n is a positive integer.

[0058] In sub-step 252, the output audio signal is obtained according to the audio sub-signals corresponding to the plurality of positions respectively.

[0059] The output audio signal can be obtained by combining the audio sub-signals corresponding to the plurality of positions respectively.

[0060] For example, the output audio signal PCM is obtained by combining the audio sub-signal PCM[0] corresponding to the first position, the audio sub-signal PCM[1] corresponding to the second position, …, and the audio sub-signal PCM[n] corresponding to the nth position, which is the composite audio signal after the synthesis of the audio subbands.

[0061] The above method realizes the equalization adjustment of the audio signal without changing the energy of the audio signal, and realizes the consistency of the auditory perception of each audio subband, thereby improving the quality of the audio equalization.

[0062] Fig. 3 shows a flowchart of an audio equalization method. First, a user inputs an audio signal and an equalization adjustment parameter, and a filter bank filters the input audio signal to obtain a plurality of audio subbands. Then, according to the at least one frequency point to be adjusted and the gain corresponding to the at least one frequency point included in the equalization adjustment parameter, the delay time corresponding to each audio subband is determined. The plurality of audio subbands are delayed according to the delay time to obtain delayed audio subbands. The delayed audio subbands are synthesized to obtain an output audio signal.

[0063] The technical scheme provided in the embodiments of the present application filters the input audio signal to obtain a plurality of audio subbands, so that the gain of the frequency point in the equalization adjustment process of the audio signal can be converted into the delay time corresponding to the audio subband, direct amplification or attenuation of the amplitude of the audio signal is avoided, and thus the influence on the energy of the audio signal is avoided, which causes the audio signal to overflow and the compression effect or the clipping distortion to occur. The equalization adjustment of the audio signal is realized while ensuring the normal output of the audio signal, and the delay time corresponding to the audio subband is calculated and each audio subband is delayed, so that the time of the auditory perception of the audio subband after the delay is unified, the auditory perception of the output audio signal at each frequency is consistent, and the quality of the audio equalization is improved. In addition, the technical method provided in the present application is easy to implement in operation, avoids modification of the hardware circuit, reduces the implementation steps of the audio equalization, saves the resources required for the audio equalization, improves the efficiency of the audio equalization, and avoids excessive cost to compensate for the defects of the audio energy.

[0064] In some embodiments, step 230 includes at least one of sub-steps 231-232.

[0065] In sub-step 231, gain values corresponding to the plurality of audio subbands are obtained according to the at least one frequency point and the gain corresponding to each of the at least one frequency point.

[0066] It should be noted that the initial gain values of the plurality of audio subbands are all 0.

[0067] In some embodiments, step 231 includes at least one of sub-steps 2311-2313.

[0068] In sub-step 2311, for a first frequency point in the at least one frequency point, a first audio subband and a second audio subband are obtained from the plurality of audio subbands, the frequency of the first frequency point is greater than or equal to the center frequency of the first audio subband, and the frequency of the first frequency point is less than the center frequency of the second audio subband.

[0069] The first frequency point is used to indicate a first position in the input audio signal that needs to adjust the signal amplitude, the frequency of the first frequency point is represented as F1, and the gain corresponding to the first frequency point is represented as G1.

[0070] The center frequency of the audio subband refers to the average value of at least one audio sample value contained in the audio subband, the first audio subband is represented as S1, the second audio subband is represented as S2, the center frequency of the first audio subband is C1, and the center frequency of the second audio subband is C2. Then C1≤F1<C2.

[0071] According to the center frequency of the first audio subband, the center frequency of the second audio subband, the frequency of the first frequency point, and the gain corresponding to the first frequency point, a sub-gain value of the first frequency point allocated to the first audio subband and a sub-gain value of the first frequency point allocated to the second audio subband are obtained.

[0072] In some embodiments, according to the center frequency of the first audio subband, the center frequency of the second audio subband, and the frequency of the first frequency point, a distance corresponding to the first audio subband is obtained; and according to the distance corresponding to the first audio subband and the gain corresponding to the first frequency point, a sub-gain value of the first frequency point allocated to the first audio subband is obtained.

[0073] According to the center frequency C1 of the first audio subband, the center frequency C2 of the second audio subband, and the frequency F1 of the first frequency point, a distance R1 corresponding to the first audio subband is obtained. According to the distance R1 corresponding to the first audio subband and the gain G1 corresponding to the first frequency point, a sub-gain value SG1 of the first frequency point allocated to the first audio subband is obtained.

[0074] For example, R1=(F1-C1) / (C2-C1), and SG1=(1-R1)*G1.

[0075] In some embodiments, according to the center frequency of the first audio subband, the center frequency of the second audio subband, and the frequency of the first frequency point, a distance corresponding to the second audio subband is obtained; and according to the distance corresponding to the second audio subband and the gain corresponding to the first frequency point, a sub-gain value of the first frequency point allocated to the second audio subband is obtained.

[0076] According to the center frequency C1 of the first audio subband, the center frequency C2 of the second audio subband, and the frequency F1 of the first frequency point, a distance R2 corresponding to the second audio subband is obtained. According to the distance R2 corresponding to the second audio subband and the gain G1 corresponding to the first frequency point, a sub-gain value SG2 of the first frequency point allocated to the second audio subband is obtained.

[0077] For example, R2=(C2-F1) / (C2-C1), and SG2=(1-R2)*G1.

[0078] The above sub-steps 2311-2312 are cycled to determine a plurality of groups of audio sub-bands meeting the frequency requirement of the first frequency point from the plurality of audio sub-bands, and to calculate the sub-gain values of the first frequency point allocated to each audio sub-band, respectively. For example, for the first frequency point, the first audio sub-band and the third audio sub-band can also be obtained from the plurality of audio sub-bands, the frequency of the first frequency point is greater than or equal to the center frequency of the first audio sub-band, and the frequency of the first frequency point is less than the center frequency of the third audio sub-band, then another sub-gain value of the first frequency point allocated to the first audio sub-band is calculated, and the sub-gain value of the first frequency point allocated to the third audio sub-band is calculated. Then, the another sub-gain value of the first frequency point allocated to the first audio sub-band in this case is added to the one sub-gain value of the first frequency point allocated to the first audio sub-band that has been obtained above, and the sub-gain values of the first frequency point allocated to the first audio sub-band are obtained by cyclically superimposing in this way.

[0079] FIG. 4 shows the process of assigning sub-gain values of the first frequency point to each audio sub-band. For the first frequency point, the first audio sub-band S1 and the second audio sub-band S2 meeting the frequency requirement are obtained from the plurality of audio sub-bands, the distance R1 corresponding to the first audio sub-band is obtained according to the center frequency of the first audio sub-band S1, the center frequency of the second audio sub-band S2, and the frequency F1 of the first frequency point, and the distance R2 corresponding to the second audio sub-band is obtained, then the sub-gain value SG1 of the first frequency point allocated to the first audio sub-band is obtained, and the sub-gain value SG2 of the first frequency point allocated to the second audio sub-band is obtained.

[0080] By respectively determining the sub-gain values of the first frequency point allocated to the first audio sub-band and the second audio sub-band, the determination method of the sub-gain values of the frequency point allocated to the audio sub-bands with different center frequencies is refined, the accuracy of the sub-gain values of each audio sub-band is improved, and thus the accuracy of the delay time of each audio sub-band is improved, and the quality of the audio equalization is improved.

[0081] In sub-step 2313, the gain value corresponding to the first audio sub-band is obtained according to the sub-gain values of the at least one frequency point allocated to the first audio sub-band, and the gain value corresponding to the second audio sub-band is obtained according to the sub-gain values of the at least one frequency point allocated to the second audio sub-band.

[0082] For example, the sub-gain values of the at least one frequency point allocated to the first audio sub-band are added to obtain the gain value corresponding to the first audio sub-band. The sub-gain values of the at least one frequency point allocated to the second audio sub-band are added to obtain the gain value corresponding to the second audio sub-band.

[0083] The gain value corresponding to each audio subband is calculated by calculating the sub-gain value of each audio subband to which the frequency point is allocated, so that the calculation of the gain value corresponding to different audio subbands is more targeted, avoiding the calculation error of the gain value corresponding to different audio subbands by using the same calculation method, improving the accuracy of the gain value corresponding to each audio subband, thereby improving the accuracy of the delay time of each audio subband and improving the quality of audio equalization.

[0084] In substep 232, the delay time corresponding to each audio subband is obtained according to the gain value corresponding to each audio subband.

[0085] According to psychoacoustics, the gain value corresponding to the audio subband is converted into the delay time corresponding to the audio subband to unify the time of the auditory perception of the audio subband, so that the auditory perception of the output audio signal on each frequency is consistent.

[0086] In some embodiments, step 232 includes at least one of substeps 2321-2323.

[0087] In substep 2321, the maximum gain value in the gain value corresponding to each audio subband is obtained.

[0088] The maximum gain value in the gain value corresponding to each audio subband is denoted as Gmax.

[0089] In substep 2322, for any audio subband, the gain value corresponding to the audio subband is subtracted from the maximum gain value to obtain the normalized gain value corresponding to the audio subband.

[0090] The gain value corresponding to each audio subband is subtracted from the maximum gain value, and the gain value of the audio subband is normalized to the maximum value of 0, i.e., the maximum value of the normalized gain value corresponding to the audio subband is 0.

[0091] In substep 2323, the delay time corresponding to the audio subband is obtained according to the center frequency and the normalized gain value of the audio subband.

[0092] According to the center frequency of the audio subband, the normalized gain value corresponding to the audio subband is converted into the delay time corresponding to the audio subband. The conversion method is as follows: for an audio subband S, the center frequency is C, and the gain is SG. The period length corresponding to the signal delay C of the audio subband is calculated for every 1 dB, and then the delay time corresponding to the audio subband S is obtained according to the number of required delay periods.

[0093] By the above manner, the gain value corresponding to the audio subband is used to obtain the delay time corresponding to the audio subband, so that the gain of the frequency point in the equalization adjustment process of the audio signal is converted into the delay time corresponding to the audio subband, thereby the time of the auditory perception of the audio subband can be unified based on the delay time corresponding to the audio subband, the auditory perception of the output audio signal at each frequency is consistent, and the equalization adjustment of the audio signal is realized. Sub-step 2323 includes the following steps.

[0094] 1. Obtain the period length of the audio subband according to the center frequency of the audio subband.

[0095] The period length of the audio subband = 1 / the center frequency of the audio subband. For example, for the audio subband S, if the center frequency of the audio subband S is 150 Hz, then the period length of the audio subband S is 1 / 150 = 0.00667 seconds.

[0096] 2. Obtain the delay period number corresponding to the normalized gain value according to the linear mapping relationship between the normalized gain value and the delay period number.

[0097] The delay period number is used to indicate the number of periods that the audio subband needs to be delayed, and there is a pre-set linear mapping relationship between the normalized gain value and the delay period number. The linear mapping relationship between the normalized gain value and the delay period number is not limited in the present application.

[0098] For example, if the normalized gain value of the audio subband S is -10 dB and the delay period number is 10, it means that the audio subband needs to be delayed for 10 periods. If the normalized gain value of the audio subband S is -20 dB and the delay period number is 20, it means that the audio subband needs to be delayed for 20 periods.

[0099] 3. Obtain the delay time corresponding to the audio subband according to the delay period number corresponding to the normalized gain value and the period length of the audio subband.

[0100] For example, the delay time corresponding to the audio subband is obtained by multiplying the delay period number corresponding to the normalized gain value by the period length of the audio subband.

[0101] For example, if the normalized gain value of the audio subband S is -10 dB and the delay period number is 10, the delay time corresponding to the audio subband is 0.0667 seconds. If the normalized gain value of the audio subband S is -20 dB and the delay period number is 20, the delay time corresponding to the audio subband is 0.1334 seconds.

[0102] In some embodiments, for the regularized gain value with a decimal, an integer can be obtained by using a method of rounding to negative infinity, and the number of delay periods corresponding to the regularized gain value can be determined according to the integer regularized gain value. For example, if the regularized gain value of the audio subband S is -10.5 dB, the integer regularized gain value obtained by rounding the regularized gain value is -11 dB, and correspondingly, the number of delay periods obtained according to the integer regularized gain value is 11.

[0103] By determining the number of delay periods required by the audio subband, the delay time corresponding to the audio subband can be obtained according to the period length of the audio subband, ensuring the normalization of the calculation process of the delay time, improving the accuracy of the delay time corresponding to the audio subband, and simplifying the calculation steps of the delay time, thereby improving the efficiency of the audio equalization.

[0104] By calculating the gain value corresponding to each audio subband by the above method and converting the gain value corresponding to the audio subband into the delay time corresponding to the audio subband, the equalization adjustment of the audio signal under the premise of not changing the energy of the audio signal is realized, the quality of the audio equalization is improved while ensuring the normal output of the audio signal, and better audio hearing effect is brought to the user.

[0105] The following is an apparatus embodiment of the present application, which can be used to execute the method embodiments of the present application. For details not disclosed in the apparatus embodiments of the present application, please refer to the method embodiments of the present application.

[0106] Please refer to FIG. 5, which shows a block diagram of an audio equalization apparatus according to an embodiment of the present application. The apparatus has the functions of implementing the above-mentioned audio equalization method, which can be implemented by hardware or corresponding software executed by hardware. The apparatus can be the computer device introduced above or can be arranged in the computer device. As shown in FIG. 5, the apparatus 500 can include a signal acquisition module 510, a signal filtering module 520, a delay determination module 530, a delay processing module 540, and a subband synthesis module 550.

[0107] The signal acquisition module 510 is configured to acquire an input audio signal and an equalization adjustment parameter, wherein the equalization adjustment parameter includes at least one frequency point to be adjusted and a gain corresponding to each of the at least one frequency point, and the gain is used to amplify the frequency of the frequency point.

[0108] The signal filtering module 520 is configured to filter the input audio signal by a filter bank to obtain a plurality of audio subbands, wherein the filter bank includes a low-pass filter, a high-pass filter, and at least one band-pass filter, and the at least one band-pass filter is a band-pass filter divided based on the critical frequency band of the Bark scale.

[0109] The delay determination module 530 is configured to obtain delay times corresponding to the plurality of audio subbands respectively according to the at least one frequency point and the gain corresponding to the at least one frequency point respectively, the delay times being used to unify the auditory perception of the audio subbands.

[0110] The delay processing module 540 is configured to perform delay processing on the plurality of audio subbands according to the delay times corresponding to the plurality of audio subbands respectively, to obtain a plurality of delayed audio subbands.

[0111] The subband synthesis module 550 is configured to synthesize the plurality of delayed audio subbands to obtain an output audio signal.

[0112] In some embodiments, the delay determination module 530 includes:

[0113] The gain determination unit is configured to obtain gain values corresponding to the plurality of audio subbands respectively according to the at least one frequency point and the gain corresponding to the at least one frequency point respectively.

[0114] The delay determination unit is configured to obtain delay times corresponding to the plurality of audio subbands respectively according to the gain values corresponding to the plurality of audio subbands respectively.

[0115] In some embodiments, the gain determination unit is configured to:

[0116] For a first frequency point in the at least one frequency point, a first audio subband and a second audio subband are obtained from the plurality of audio subbands, the frequency of the first frequency point being greater than or equal to the center frequency of the first audio subband, and the frequency of the first frequency point being less than the center frequency of the second audio subband.

[0117] According to the center frequency of the first audio subband, the center frequency of the second audio subband, the frequency of the first frequency point, and the gain corresponding to the first frequency point, a sub-gain value of the first frequency point allocated to the first audio subband and a sub-gain value of the first frequency point allocated to the second audio subband are obtained.

[0118] According to the sub-gain values of the at least one frequency point allocated to the first audio subband respectively, a gain value corresponding to the first audio subband is obtained, and according to the sub-gain values of the at least one frequency point allocated to the second audio subband respectively, a gain value corresponding to the second audio subband is obtained.

[0119] In some embodiments, the gain determination unit is configured to:

[0120] obtaining a distance corresponding to the first audio subband and a distance corresponding to the second audio subband according to the center frequency of the first audio subband, the center frequency of the second audio subband and the frequency of the first frequency point;

[0121] obtaining a sub-gain value of the first frequency point allocated to the first audio subband according to the distance corresponding to the first audio subband and the gain corresponding to the first frequency point;

[0122] obtaining a sub-gain value of the first frequency point allocated to the second audio subband according to the distance corresponding to the second audio subband and the gain corresponding to the first frequency point.

[0123] In some embodiments, the delay determination unit is configured to:

[0124] obtaining a maximum gain value in the gain values corresponding to the plurality of audio subbands respectively;

[0125] obtaining a normalized gain value corresponding to the audio subband by subtracting the maximum gain value from the gain value corresponding to the audio subband for any one of the audio subbands;

[0126] obtaining a delay time corresponding to the audio subband according to the center frequency of the audio subband and the normalized gain value.

[0127] In some embodiments, the delay determination unit is configured to:

[0128] obtaining a period length of the audio subband according to the center frequency of the audio subband;

[0129] obtaining a delay period number corresponding to the normalized gain value according to a linear mapping relationship between the normalized gain value and the delay period number;

[0130] obtaining a delay time corresponding to the audio subband according to the delay period number corresponding to the normalized gain value and the period length of the audio subband.

[0131] In some embodiments, the subband synthesis module 550 is configured to:

[0132] adding audio sample values respectively at the same position in the plurality of audio subbands after the delay to obtain audio sub-signals corresponding to a plurality of positions respectively;

[0133] obtaining the output audio signal according to the audio sub-signals corresponding to the plurality of positions respectively.

[0134] It should be noted that the apparatus provided by the above embodiments, in realizing its functions, only divides the above-mentioned various functional modules by way of example, and in actual application, the above-mentioned functions can be completed by different functional modules according to needs, that is, the content structure of the device is divided into different functional modules to complete all or part of the functions described above. In addition, the apparatus and method embodiments provided by the above embodiments belong to the same concept, and the specific implementation process is detailed in the method embodiments, which will not be repeated here.

[0135] Please refer to FIG. 6, which shows a structural block diagram of a computer device 600 provided by an embodiment of the present application. The computer device 600 can be any electronic device with data computing, processing and storage functions. The computer device 600 can be used to implement the audio equalization method provided in the above embodiments.

[0136] Generally, the computer device 600 includes a processor 601 and a memory 602.

[0137] The processor 601 can include one or more processing cores, such as a 4-core processor, an 8-core processor, etc. The processor 601 can be implemented in at least one of the hardware forms of DSP (Digital Signal Processing), FPGA (Field Programmable Gate Array), and PLA (Programmable Logic Array). The processor 601 can also include a main processor and a coprocessor. The main processor is a processor for processing data in an awake state, also known as a CPU (Central Processing Unit). The coprocessor is a low-power processor for processing data in a standby state. In some embodiments, the processor 601 can be integrated with a GPU (Graphics Processing Unit) that is responsible for rendering and drawing the content to be displayed on the display screen. In some embodiments, the processor 601 can also include an AI processor for processing machine learning-related computing operations.

[0138] The memory 602 can include one or more computer-readable storage media, which can be non-transitory. The memory 602 can also include a high-speed random access memory and a non-volatile memory such as one or more disk storage devices, flash storage devices. In some embodiments, the non-transitory computer-readable storage medium in the memory 602 is used to store a computer program configured to be executed by one or more processors to implement the above-mentioned audio equalization method.

[0139] Those skilled in the art can understand that the structure shown in FIG. 6 does not constitute a limitation on the computer device 600, and can include more or fewer components than shown, or combine certain components, or adopt a different arrangement of components.

[0140] In the illustrative embodiment, a computer readable storage medium is also provided, in which a computer program is stored, the computer program, when executed by a processor of a computer device, implements the above-mentioned audio equalization method. Optionally, the above-mentioned computer readable storage medium can be a ROM (Read-Only Memory), a RAM (Random Access Memory), a CD-ROM (Compact Disc Read-Only Memory), a magnetic tape, a floppy disk, an optical data storage device, etc.

[0141] In the illustrative embodiment, a computer program product is also provided, which includes a computer program stored in a computer readable storage medium. The processor of a computer device reads the computer program from the computer readable storage medium, and the processor executes the computer program, so that the computer device executes the above-mentioned audio equalization method.

[0142] It should be understood that "multiple" referred to herein means two or more. "And / or", which describes the association relationship of associated objects, means that there can be three relationships, for example, A and / or B can mean that A exists alone, A and B exist together, and B exists alone. The character " / " generally represents that the associated objects before and after it are in an "or" relationship. In addition, the step numbers described herein only exemplarily show a possible execution order between steps, and in some other embodiments, the above steps can also be executed in a different order from the numbering order, such as simultaneously executing two steps with different numbers, or executing two steps with different numbers in an order opposite to the illustration, and the embodiments of the present application do not limit this.

[0143] The above only describes exemplary embodiments of the present application and does not limit the present application. Any modification, equivalent replacement, improvement, etc. made within the spirit and principles of the present application shall be included in the protection scope of the present application.

Claims

1. An audio equalization method, the method being performed by a computer device, the method comprising: obtaining an input audio signal and equalization adjustment parameters, the equalization adjustment parameters comprising at least one frequency point to be adjusted and a gain corresponding to each of the at least one frequency point, the gain being used to amplify the frequency of the frequency point; filtering the input audio signal through a filter bank to obtain a plurality of audio subbands, the filter bank comprising a low-pass filter, a high-pass filter, and at least one band-pass filter, the at least one band-pass filter being a band-pass filter divided based on critical bands of Bark scale; obtaining a delay time corresponding to each of the plurality of audio subbands according to the at least one frequency point and the gain corresponding to each of the at least one frequency point, the delay time being used to unify the auditory perception of the audio subbands; delay processing the plurality of audio subbands according to the delay time corresponding to each of the plurality of audio subbands to obtain a plurality of delayed audio subbands; combining the plurality of delayed audio subbands to obtain an output audio signal.

2. The method of claim 1, wherein, The obtaining of the delay time corresponding to each of the plurality of audio subbands according to the at least one frequency point and the gain corresponding to each of the at least one frequency point comprises: obtaining a gain value corresponding to each of the plurality of audio subbands according to the at least one frequency point and the gain corresponding to each of the at least one frequency point; obtaining the delay time corresponding to each of the plurality of audio subbands according to the gain value corresponding to each of the plurality of audio subbands.

3. The method of claim 2, wherein, The obtaining of the gain value corresponding to each of the plurality of audio subbands according to the at least one frequency point and the gain corresponding to each of the at least one frequency point comprises: for a first frequency point of the at least one frequency point, obtaining a first audio subband and a second audio subband from the plurality of audio subbands, the frequency of the first frequency point being greater than or equal to a center frequency of the first audio subband and less than a center frequency of the second audio subband; obtaining a sub-gain value of the first frequency point allocated to the first audio subband and a sub-gain value of the first frequency point allocated to the second audio subband according to the center frequency of the first audio subband, the center frequency of the second audio subband, the frequency of the first frequency point, and the gain corresponding to the first frequency point; obtaining a gain value corresponding to the first audio subband according to the sub-gain value of the first frequency point allocated to the first audio subband, and obtaining a gain value corresponding to the second audio subband according to the sub-gain value of the first frequency point allocated to the second audio subband.

4. The method of claim 3, wherein, The obtaining of the sub-gain value of the first frequency point allocated to the first audio subband and the sub-gain value of the first frequency point allocated to the second audio subband according to the center frequency of the first audio subband, the center frequency of the second audio subband, the frequency of the first frequency point, and the gain corresponding to the first frequency point comprises: obtaining a distance corresponding to the first audio subband and a distance corresponding to the second audio subband according to the center frequency of the first audio subband, the center frequency of the second audio subband and the frequency of the first frequency point; obtaining a sub-gain value of the first frequency point allocated to the first audio subband according to the distance corresponding to the first audio subband and the gain corresponding to the first frequency point; obtaining a sub-gain value of the first frequency point allocated to the second audio subband according to the distance corresponding to the second audio subband and the gain corresponding to the first frequency point.

5. The method according to any one of claims 2 to 4, wherein, The method further includes: obtaining a maximum gain value in the gain values corresponding to the plurality of audio subbands respectively; for any one of the audio subbands, obtaining a normalized gain value corresponding to the audio subband by subtracting the maximum gain value from the gain value corresponding to the audio subband; obtaining a delay time corresponding to the audio subband according to the center frequency of the audio subband and the normalized gain value.

6. The method of claim 5, wherein, The method further includes: obtaining a period length of the audio subband according to the center frequency of the audio subband; obtaining a delay period number corresponding to the normalized gain value according to a linear mapping relationship between the normalized gain value and the delay period number; obtaining a delay time corresponding to the audio subband according to the delay period number corresponding to the normalized gain value and the period length of the audio subband.

7. The method according to any one of claims 1 to 6, wherein, The method further includes: adding audio sample values respectively at the same position in the plurality of audio subbands after the delay to obtain an audio sub-signal corresponding to each position; obtaining the output audio signal according to the audio sub-signals corresponding to the plurality of positions respectively. 8.An audio equalization device, the device comprising: a signal obtaining module configured to obtain an input audio signal and an equalization adjustment parameter, the equalization adjustment parameter comprising at least one frequency point to be adjusted and a gain corresponding to the at least one frequency point respectively, the gain being used to amplify the frequency of the frequency point; a signal filtering module configured to filter the input audio signal through a filter bank to obtain a plurality of audio subbands, the filter bank comprising a low-pass filter, a high-pass filter and at least one band-pass filter, the at least one band-pass filter being a band-pass filter based on a critical band of Bark scale; a delay determining module configured to obtain a delay time corresponding to the plurality of audio subbands respectively according to the at least one frequency point and the gain corresponding to the at least one frequency point respectively, the delay time being a time for unifying the auditory perception of the audio subbands; a delay processing module configured to delay the plurality of audio subbands according to the delay time corresponding to the plurality of audio subbands respectively to obtain a plurality of audio subbands after the delay; a sub-band synthesizing module configured to synthesize the plurality of audio subbands after the delay to obtain an output audio signal.

9. A computer device comprising a processor and a memory, the memory having stored therein a computer program, the computer program being loaded and executed by the processor to implement the audio equalization method according to any one of claims 1 to 7.

10. A computer readable storage medium having stored therein a computer program, the computer program being loaded and executed by a processor to implement the audio equalization method according to any one of claims 1 to 7.

11. A computer program product comprising a computer program, the computer program being loaded and executed by a processor to implement the audio equalization method according to any one of claims 1 to 7.

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