Audio processing system, method and equipment and storage medium

Abstract

The invention relates to the field of data processing, and discloses an audio processing system, method and device and a storage medium. The system comprises a detection module used for detecting the volume of an original digital audio signal in real time and triggering a healthy listening mode when the volume is smaller than a first preset sound pressure level threshold value; the initialization module is used for initializing the dynamic range expansion processor and the audio compensation processor; the expansion module is used for carrying out dynamic range expansion on the original digital audio signal by utilizing a dynamic range expansion processor to obtain an expanded digital audio signal; the compensation module is used for performing loudness compensation on the expanded digital audio signal by using an audio compensation processor to obtain a compensated digital audio signal; and the conversion module is used for converting the compensated digital audio signal into an analog audio signal and playing the analog audio signal. According to the invention, through full-process cooperation of detection, initialization, expansion, compensation and conversion modules, hearing protection and high-quality audition experience are taken into account.

Description

Technical Field

[0001] This invention relates to the field of data processing, and more particularly to an audio processing system, method, apparatus, and storage medium. Background Technology

[0002] With the increasing popularity of audio devices (such as headphones and smart speakers), users are spending significantly more time using audio devices every day, and hearing health is gradually becoming a core concern for consumers and the industry.

[0003] However, existing audio equipment struggles to address the shortcomings in sound quality at low volumes, which can lead users to increase the volume in pursuit of a more satisfying listening experience and greater audio detail, thus hindering hearing protection. Summary of the Invention

[0004] In view of this, the purpose of the present invention is to overcome the shortcomings of the prior art and provide an audio processing system, method, device and storage medium.

[0005] This invention provides the following technical solution: In a first aspect, the present invention provides an audio processing system, the system comprising: The detection module is used to detect the volume of the original digital audio signal in real time. When the volume of the original digital audio signal is less than a first preset sound pressure level threshold, the healthy listening mode is triggered. An initialization module is used to read pre-stored user parameters after entering the healthy listening mode, and initialize the dynamic range extension processor and audio compensation processor based on the user parameters; An extension module is used to extend the dynamic range of the original digital audio signal using the dynamic range extension processor to obtain an extended digital audio signal; The compensation module is used to perform loudness compensation on the low-frequency and high-frequency bands of the extended digital audio signal using the audio compensation processor to obtain the compensated digital audio signal. The conversion module is used to convert the compensated digital audio signal into an analog audio signal and play the analog audio signal according to a preset output standard.

[0006] In an optional implementation, the extension module includes: The signal analysis unit is used to calculate the absolute value of the original digital audio signal, calculate the peak value of the original digital audio signal based on the absolute value, and calculate the root mean square value of the original digital audio signal. An adaptive parameter adjustment unit is used to determine the transient factor of the original digital audio signal based on the root mean square value and peak value of the original digital audio signal, and to determine the target spread ratio based on the transient factor.

[0007] In an optional implementation, the adaptive parameter adjustment unit includes: The normalization subunit is used to linearly convert the original digital audio signal whose root mean square value is within a preset sound pressure level range to a preset numerical range to obtain the converted digital audio signal. A transient detection subunit is used to determine whether the converted digital audio signal is a transient signal based on the difference between the peak value and the root mean square value. When the difference is greater than a first preset transient threshold, the converted digital audio signal is identified as a strong transient signal. When the difference is less than or equal to the first preset transient threshold and greater than a second preset transient threshold, the converted digital audio signal is identified as a weak transient signal. When the difference is less than or equal to the second preset transient threshold, the converted digital audio signal is identified as having no transient signal. The unit also determines the transient factor of the converted digital audio signal and determines the target spread ratio based on the transient factor.

[0008] In an optional implementation, the extension module further includes: An envelope detection unit is used to calculate the absolute value of the converted digital audio signal and the current envelope value, and to calculate the target envelope value based on the absolute value of the converted digital audio signal and the current envelope value. The gain calculation unit is used to calculate the target gain using the target spread ratio and the target envelope value, and update the original digital audio signal according to the target gain to obtain the expanded digital audio signal.

[0009] In an optional implementation, the envelope detection unit includes: The mode selection subunit is used to enter attack mode when the absolute value of the converted digital audio signal is greater than the current envelope value, and to enter release mode when the absolute value of the converted digital audio signal is less than the current envelope value. An envelope update subunit is used to calculate the target envelope value of the converted digital audio signal using a preset attack coefficient, the absolute value, and the current envelope value in the attack mode, and to calculate the target envelope value of the converted digital audio signal using a preset release coefficient, the absolute value, and the current envelope value in the release mode.

[0010] In an optional implementation, the gain calculation unit includes: The soft inflection point processing subunit is used to determine the upper half inflection point and the lower half inflection point using a second preset sound pressure level threshold. When the sound pressure level of the converted digital audio signal is greater than the upper half inflection point, the converted digital audio signal is directly used as the digital audio signal after soft inflection point processing. When the sound pressure level of the converted digital audio signal is less than or equal to the upper half inflection point and greater than or equal to the lower half inflection point, the converted digital audio signal is smoothed using a quadratic curve to obtain the digital audio signal after soft inflection point processing. When the sound pressure level of the converted digital audio signal is less than the lower half inflection point, the converted digital audio signal is fixed to a preset gain value to obtain the digital audio signal after soft inflection point processing. The gain calculation subunit is used to calculate the target gain based on the digital audio signal after the soft inflection point processing, and using a preset dynamic compensation threshold, the target spread ratio and the target envelope value. The gain update subunit is used to determine the smoothing coefficients based on the attack mode or release mode of the converted digital audio signal, calculate the update gain based on the target gain and the smoothing coefficients, and update the original digital audio signal using the update gain to obtain the extended digital audio signal.

[0011] In an optional implementation, the compensation module includes: A low-shelf filter is used to perform gain compensation on frequency bands less than a first preset frequency in the extended digital audio signal based on the perceived attenuation characteristics of the low-frequency band in the preset equal loudness curve. A high-shelf filter is used to perform gain compensation on frequency bands above a second preset frequency in the extended digital audio signal based on the perceived attenuation characteristics of the high-frequency band in the preset equal loudness curve.

[0012] In a second aspect, the present invention provides an audio processing method, the method comprising: The volume of the original digital audio signal is detected in real time. When the volume of the original digital audio signal is less than a first preset sound pressure level threshold, a healthy listening mode is triggered. After entering the healthy listening mode, the pre-stored user parameters are read, and the dynamic range extension processor and audio compensation processor are initialized based on the user parameters; The original digital audio signal is dynamically extended using the dynamic range extender to obtain an extended digital audio signal. The audio compensation processor is used to perform loudness compensation on the low-frequency and high-frequency bands of the extended digital audio signal to obtain the compensated digital audio signal. The compensated digital audio signal is converted into an analog audio signal, and the analog audio signal is played according to a preset output standard.

[0013] Thirdly, the present invention provides a computer device including a memory and a processor, wherein the memory stores a computer program, and the processor executes the computer program to implement the steps of the audio processing method described in the second aspect above.

[0014] Fourthly, the present invention provides a computer-readable storage medium storing a computer program that, when executed by a processor, implements the steps of the audio processing method described in the second aspect above.

[0015] The beneficial effects of this application are: The audio processing method provided in this application embodiment, through the coordinated process of "detection-initialization-dynamic expansion-frequency compensation-conversion output", defines the effective scenarios for subsequent processing by the detection module, provides the parameter basis for the core processor by the initialization module, optimizes the dynamic range of the signal by the expansion module, corrects the frequency response by the compensation module, and ensures accurate signal output by the conversion module. This not only solves the problems of loss of details in weak signals and insufficient perception of high and low frequencies at low volumes, but also avoids transient distortion and abrupt gain changes. The final output is audio with reasonable dynamics, frequency adaptation, and safe volume. While ensuring that details can be heard clearly, it reduces the need for users to increase the volume due to poor listening experience, achieving the dual goals of "natural listening experience + hearing protection".

[0016] To make the above-mentioned objects, features and advantages of the present invention more apparent and understandable, preferred embodiments are described below in detail with reference to the accompanying drawings. Attached Figure Description

[0017] To more clearly illustrate the technical solutions of the embodiments of the present invention, the accompanying drawings used in the embodiments will be briefly described below. It should be understood that the following drawings only show some embodiments of the present invention and should not be regarded as a limitation on the scope. For those skilled in the art, other related drawings can be obtained based on these drawings without creative effort. In the various drawings, similar components are numbered similarly.

[0018] Figure 1 A schematic diagram of an acoustic standard equal loudness level curve provided in an embodiment of this application is shown; Figure 2 This paper shows a schematic diagram of the structure of an audio processing system provided in an embodiment of this application; Figure 3 A flowchart of an audio processing method provided in an embodiment of this application is shown; Figure 4 A schematic diagram of the structure of a computer device provided in an embodiment of this application is shown. Detailed Implementation

[0019] Embodiments of the present invention are described in detail below. Examples of these embodiments are shown in the accompanying drawings, wherein the same or similar reference numerals denote the same or similar elements or elements having the same or similar functions throughout. The embodiments described below with reference to the accompanying drawings are exemplary and are only used to explain the present invention, and should not be construed as limiting the present invention.

[0020] It should be noted that the terms "first" and "second" are used for descriptive purposes only and should not be construed as indicating or implying relative importance or implicitly specifying the number of technical features indicated. Therefore, a feature defined as "first" or "second" may explicitly or implicitly include one or more of that feature. In the description of this invention, "a plurality of" means two or more, unless otherwise explicitly specified.

[0021] Unless otherwise defined, all technical and scientific terms used herein have the same meaning as commonly understood by one of ordinary skill in the art to which this application belongs. The terminology used herein in the template description is for the purpose of describing particular embodiments only and is not intended to limit the invention. The term "and / or" as used herein includes any and all combinations of one or more of the associated listed items.

[0022] Example 1 First, it needs to be explained that, as Figure 1 The image shows the acoustic standard equal-loudness level curves (ISO 226:2003). These curves describe the differences in sound pressure level (SPL) required for different frequencies to perceive the same loudness. The horizontal axis of the equal-loudness level curves represents frequency (Hz), and the vertical axis represents sound pressure level (dB SPL). Each curve is labeled with a loudness level (Phon), the value of which is equal to the SPL at 1kHz. For example, the 20Phon curve indicates that a 1kHz sound is perceived as 20Phon at 20dB SPL, while a 100Hz sound requires an increase to 50dB SPL to achieve the same loudness.

[0023] The physiological basis for this is that the human ear is most sensitive to mid-frequency (1–4kHz) frequencies (overlapping with the speech frequency range), while the perception of low frequencies (<200Hz) and high frequencies (>8kHz) decreases significantly as the volume decreases. This is because the cochlear basilar membrane structure is more efficient in responding to mid-frequency vibrations, and the resonant frequency of the external auditory canal is approximately 3kHz.

[0024] See Table 1 below for the perceived difference between volume and frequency, combined with... Figure 1 It can be seen that at low volume (<40Phon): the perception of low / high frequencies drops sharply, the curve is "U-shaped", and musical details (such as bass dynamics and cymbal overtones) are lost; at high volume (>80Phon): the curve tends to be flat, the perception of low and high frequencies is enhanced, and the details are richer.

[0025] Table 1. Perceived differences in volume and frequency

[0026] like Figure 2 The diagram shown is a structural schematic of an audio processing system 100 according to an embodiment of this application. The system includes a detection module 110, an initialization module 120, an expansion module 130, a compensation module 140, and a conversion module 150.

[0027] Preferably, the detection module 110 is used to detect the volume of the original digital audio signal in real time, and triggers the healthy listening mode when the volume of the original digital audio signal is less than a first preset sound pressure level threshold.

[0028] Understandably, the detection module 110 is responsible for detecting the volume of the input raw digital audio signal. When the volume of the raw digital audio signal is less than the first preset sound pressure level threshold (e.g., -60dB), the healthy listening mode is automatically triggered.

[0029] It should be noted that there are two ways to activate the Healthy Listening Mode: 1) Directly trigger the Healthy Listening Mode via a dedicated physical button or touch button. Once triggered, it bypasses the digital volume preamp and forcibly locks the output of subsequent conversion modules to a preset output standard (e.g., 0dB); 2) Trigger the Healthy Listening Mode by reading a pre-stored user profile. In Healthy Listening Mode, system volume adjustment only affects the digital attenuation of the preamp (e.g., adjustment range of -100dB to 0dB), which avoids distortion of the analog signal in the subsequent stage and ensures that the compensation algorithm always operates within the dynamic range of the original digital audio signal, providing a "clean" signal foundation for subsequent processing.

[0030] Preferably, the initialization module 120 is used to read pre-stored user parameters after entering the healthy listening mode, and initialize the dynamic range extension processor and the audio compensation processor based on the user parameters.

[0031] Understandably, after the healthy listening mode is activated, the system enters the initialization phase. The initialization module 120 reads the pre-stored user parameters, such as dynamic compensation threshold, filter gain, trigger time, release time, smoothness, etc. The parameters can be customized by the user.

[0032] The initialization module 120 then initializes the two core processors based on user parameters: the dynamic range extension processor (responsible for optimizing weak signal details) and the audio compensation processor (responsible for high and low frequency loudness compensation). At the same time, it loads preset filter parameters (such as the frequency points, gain, and Q values ​​of low-shelf filters and high-shelf filters) to ensure that subsequent processing "has parameters to follow".

[0033] Preferably, the expansion module 130 is used to perform dynamic range expansion on the original digital audio signal using the dynamic range expansion processor to obtain an expanded digital audio signal.

[0034] In one alternative implementation, the expansion module 130 includes a signal analysis unit 131 and an adaptive parameter adjustment unit 132.

[0035] Optionally, the signal analysis unit 131 is used to calculate the absolute value of the original digital audio signal, calculate the peak value of the original digital audio signal based on the absolute value, and calculate the root mean square value of the original digital audio signal.

[0036] Understandably, the signal analysis unit 131 first calculates the absolute value of the original digital audio signal, calculates the peak value of the original digital audio signal based on the absolute value, and then converts it into sound pressure level. The formula is: peak value dB SPL = 20 × log10 (abs_sample), where abs_sample represents the absolute value of the original digital audio signal (eliminating the influence of positive and negative signal directions), and log10 is the logarithm to the base 10. This formula can convert the amplitude value of the original digital audio signal into a sound pressure level that can be perceived by the human ear.

[0037] Next, within a specified time window (preset window_ms=50ms), the samples are squared and summed, and then averaged to calculate the root mean square (RMS) value of the original digital audio signal. This value is then converted to sound pressure level (SPL) using the formula: RMS value dB SPL=20×log10(sqrt(rms_sum / rms_count)), where rms_sum is the sum of squares of all original digital audio signals within the specified time window (squaring amplifies differences in strong signals and eliminates positive and negative effects), rms_count is the total number of original digital audio signals within the specified time window, and sqrt(rms_sum / rms_count) is the RMS value (reflecting the average intensity of the original digital audio signal within the specified time window).

[0038] Optionally, the adaptive parameter adjustment unit 132 is used to determine the transient factor of the original digital audio signal based on the root mean square value and peak value of the original digital audio signal, and to determine the target spread ratio based on the transient factor.

[0039] In one alternative implementation, the adaptive parameter adjustment unit 132 includes a normalization subunit 1321 and a transient detection subunit 1322.

[0040] The normalization subunit 1321 is used to linearly convert the original digital audio signal whose root mean square value is within a preset sound pressure level range to a preset numerical range to obtain the converted digital audio signal.

[0041] Understandably, the normalization subunit 1321 is responsible for linearly transforming the original digital audio signal with a root mean square value within a preset sound pressure level range (e.g., -20dB to -100dB) to a preset numerical range (e.g., 0 to 1), obtaining the transformed digital audio signal. The purpose of the normalization subunit 1321 is to generate a "non-linear expansion ratio curve": the closer the value is to 0 (corresponding to a weaker original digital audio signal, such as -144dB, a silent signal), the higher the expansion ratio (enhancing weak signals); the closer the value is to 1 (corresponding to a stronger original digital audio signal, such as 0dB, the maximum signal), the lower the expansion ratio (avoiding excessive amplification of strong signals), providing adaptation parameters for subsequent gain calculations.

[0042] The transient detection subunit 1322 is used to determine whether the converted digital audio signal is a transient signal based on the difference between the peak value and the root mean square value. When the difference is greater than a first preset transient threshold, the converted digital audio signal is identified as a strong transient signal. When the difference is less than or equal to the first preset transient threshold and greater than a second preset transient threshold, the converted digital audio signal is identified as a weak transient signal. When the difference is less than or equal to the second preset transient threshold, the converted digital audio signal is identified as having no transient signal. The transient factor of the converted digital audio signal is determined, and the target spread ratio is determined based on the transient factor.

[0043] Understandably, the transient detection subunit 1322 is responsible for determining whether the converted digital audio signal is a transient signal (such as drum beats or instrument loudspeakers) by the difference between the peak value and the root mean square value, and adjusting the spread ratio to avoid the transient signal being over-spread, which could lead to distortion or a "sucking" effect. Specifically, the behavior is as follows: 1) When the difference is greater than the first preset transient threshold (e.g., 12dB), the converted digital audio signal is identified as a strong transient signal. At this time, the transient factor can be 0.85, and the target expansion ratio = original expansion ratio × 0.85 (reducing the expansion ratio and reducing the amplification of strong transients); 2) When the difference is less than or equal to the first preset transient threshold (e.g., 12dB) and greater than the second preset transient threshold (e.g., 6dB), the converted digital audio signal is identified as a weak transient signal. At this time, the transient factor can be 0.95, and the target expansion ratio = original expansion ratio × 0.95 (slightly reducing the expansion ratio); 3) When the difference is less than or equal to the second preset transient threshold, the converted digital audio signal is identified as a signal without transients. At this time, the transient factor can be 1.0, and the target expansion ratio = original expansion ratio × 1.0, that is, maintaining the original expansion ratio (normal amplification of weak signals).

[0044] In an optional implementation, the expansion module 130 further includes an envelope detection unit 133 and a gain calculation unit 134.

[0045] Optionally, the envelope detection unit 133 is used to calculate the absolute value of the converted digital audio signal and the current envelope value, and to calculate the target envelope value based on the absolute value of the converted digital audio signal and the current envelope value.

[0046] In one alternative implementation, the envelope detection unit 133 includes a mode selection subunit 1331 and an envelope update subunit 1332.

[0047] The mode selection subunit 1331 is used to enter the attack mode when the absolute value of the converted digital audio signal is greater than the current envelope value, and to enter the release mode when the absolute value of the converted digital audio signal is less than the current envelope value.

[0048] Understandably, the mode selection subunit 1331 is responsible for comparing the absolute value of the converted digital audio signal with the current envelope value. When the absolute value of the converted digital audio signal is greater than the current envelope value (signal amplitude increases, such as when an instrument starts playing), it enters the attack mode (rapidly tracking the increase in amplitude). When the absolute value of the converted digital audio signal is less than the current envelope value (signal amplitude decreases, such as when an instrument decays), it enters the release mode (slowly tracking the decrease in amplitude).

[0049] Envelope update subunit 1332 is used to calculate the target envelope value of the converted digital audio signal using a preset attack coefficient, the absolute value and the current envelope value in the attack mode, and to calculate the target envelope value of the converted digital audio signal using a preset release coefficient, the absolute value and the current envelope value in the release mode.

[0050] Understandably, in attack mode, the formula for calculating the target envelope value is:

[0051] In release mode, the formula for calculating the target envelope value is:

[0052] in, Indicates the target envelope value. Indicates the current envelope value. This indicates the preset attack coefficient. Indicates the preset release coefficient. This represents the absolute value. The two formulas above allow the envelope value to smoothly follow changes in signal amplitude, avoiding abrupt changes.

[0053] Optionally, the gain calculation unit 134 is used to calculate the target gain using the target spread ratio and the target envelope value, and update the original digital audio signal according to the target gain to obtain the expanded digital audio signal.

[0054] In one alternative implementation, the gain calculation unit 134 includes a soft inflection point processing subunit 1341, a gain calculation subunit 1342, and a gain update subunit 1343.

[0055] The soft inflection point processing subunit 1341 is used to determine the upper half inflection point and the lower half inflection point using a second preset sound pressure level threshold. When the sound pressure level of the converted digital audio signal is greater than the upper half inflection point, the converted digital audio signal is directly used as the digital audio signal after soft inflection point processing. When the sound pressure level of the converted digital audio signal is less than or equal to the upper half inflection point and greater than or equal to the lower half inflection point, the converted digital audio signal is smoothed using a quadratic curve to obtain the digital audio signal after soft inflection point processing. When the sound pressure level of the converted digital audio signal is less than the lower half inflection point, the converted digital audio signal is fixed to a preset gain value to obtain the digital audio signal after soft inflection point processing.

[0056] Understandably, the soft inflection point processing subunit 1341 is responsible for segmenting the signal to solve the problem of "sudden signal gain change near the threshold". First, the upper half inflection point and the lower half inflection point are determined using the second preset sound pressure level threshold and smoothness. For example, the second preset sound pressure level threshold is -50dB, the smoothness is 6dB, and the half inflection point is 6÷2=3dB. Therefore, the upper half inflection point is -50dB+3dB=-47dB, and the lower half inflection point is -50dB-3dB=-53dB.

[0057] When the sound pressure level of the converted digital audio signal is greater than the upper half-inflection point, the converted digital audio signal is directly used as the digital audio signal after soft inflection point processing without additional adjustment; when the sound pressure level of the converted digital audio signal is less than or equal to the upper half-inflection point and greater than or equal to the lower half-inflection point, a quadratic curve is used to smooth the converted digital audio signal before it is used as the digital audio signal after soft inflection point processing to avoid gain jumps; when the sound pressure level of the converted digital audio signal is less than the lower half-inflection point, the converted digital audio signal is fixed to a preset gain value before it is used as the digital audio signal after soft inflection point processing to ensure stable amplification of weak signals.

[0058] The gain calculation subunit 1342 is used to calculate the target gain based on the digital audio signal after the soft inflection point processing, and using a preset dynamic compensation threshold, the target spread ratio and the target envelope value.

[0059] Understandably, based on the digital audio signal after soft inflection processing, the target gain to be amplified is calculated using the following formula:

[0060] in, Indicates the target gain. This indicates the second preset sound pressure level threshold. This represents the target envelope value of the digital audio signal after soft inflection processing. This represents the target extension ratio. The logic of this formula is: the weaker the signal (the smaller the processed_db) and the higher the target extension ratio (the larger the adaptive_ratio), the greater the target gain. In this embodiment, the maximum target gain is limited to 12dB to avoid excessive amplification leading to distortion.

[0061] The gain update subunit 1343 is used to determine the smoothing coefficients according to the attack mode or release mode of the converted digital audio signal, calculate the update gain according to the target gain and the smoothing coefficients, and update the original digital audio signal using the update gain to obtain the extended digital audio signal.

[0062] Understandably, the formula for calculating the updated gain is as follows:

[0063] in, Indicates the update gain. Indicates the original gain. This represents the smoothing coefficient, which is applied when the converted digital audio signal is in attack mode. The attack coefficient (e.g., 0.9, corresponding to 30ms) is used to quickly update the gain to the target value. This occurs when the converted digital audio signal is in release mode. The gain is updated slowly using a release factor (e.g., 0.5, corresponding to 200ms). This formula allows the gain to transition smoothly to the target value, resulting in a more natural sound.

[0064] Next, the calculated update gain is applied to the original digital audio signal to obtain the expanded digital audio signal. The calculation formula is as follows:

[0065] in, This represents the expanded digital audio signal output. This represents the original digital audio signal. This step performs dynamic optimization, including "weak signal enhancement, transient protection, and gain smoothing," providing a signal with a "reasonable dynamic range" for subsequent high and low frequency compensation.

[0066] Preferably, the compensation module 140 is used to perform loudness compensation on the low-frequency and high-frequency bands of the extended digital audio signal using the audio compensation processor to obtain the compensated digital audio signal.

[0067] In one optional implementation, the compensation module 140 includes a low-shelf filter 141 and a high-shelf filter 142. The low-shelf filter 141 is used to perform gain compensation on frequency bands in the extended digital audio signal below a first preset frequency, based on the perceived attenuation characteristics of the low-frequency band in a preset equal-loudness curve. The high-shelf filter 142 is used to perform gain compensation on frequency bands in the extended digital audio signal above a second preset frequency, based on the perceived attenuation characteristics of the high-frequency band in the preset equal-loudness curve.

[0068] Understandably, the compensation module 140 is responsible for frequency compensation of the expanded digital audio signal output by the expansion module, solving the problem of "thin sound at low volumes," and forming a dual compensation mechanism of "intensity optimization + frequency optimization" with dynamic processing. The high and low shelf filters work synchronously to accurately match the low-volume perception characteristics of the equal loudness curve, ultimately obtaining the compensated digital audio signal.

[0069] First, an IIR (Infinite Impulse Response) filter is used to implement low-shelf filtering. Based on a preset equal-loudness curve, gain compensation is performed only on the low-frequency band below a first preset frequency (e.g., 200Hz) in the extended digital audio signal. In this embodiment, the parameters of the low-shelf filter are: cutoff frequency = 80Hz (only boosting low frequencies at 80Hz and below, matching the low-frequency attenuation band perceived by the human ear); gain = +6.0dB (boosting the low-frequency sound pressure level according to the requirements of the equal-loudness curve to compensate for perceptual defects); Q value = 0.707 (controlling the steepness of the filter curve to avoid excessive muddiness in low frequencies).

[0070] Similarly, an IIR filter is used to implement high-shelf filtering. Based on a preset equal loudness curve, gain compensation is only performed on the high-frequency band above the second preset frequency (e.g., 5kHz) in the extended digital audio signal. In this embodiment, the parameters of the high-shelf filter are: cutoff frequency = 6000Hz (only boosting high frequencies of 6000Hz and above to match the high-frequency attenuation band perceived by the human ear); gain = +2.0dB (moderately boosting the high-frequency sound pressure level to restore instrument overtones and vocal details, avoiding harshness); Q value = 1.0 (controlling the steepness of the filter curve to ensure a natural high-frequency transition).

[0071] Preferably, the conversion module 150 is used to convert the compensated digital audio signal into an analog audio signal and play the analog audio signal according to a preset output standard.

[0072] Understandably, after the "dynamic range extension" and "high and low frequency compensation" processing, the system sends the compensated digital audio signal to the conversion module 150. The conversion module 150 converts the compensated digital audio signal into an analog audio signal and plays it according to a pre-locked preset output standard (e.g., 0dB). The conversion module 150 is the endpoint of the entire processing flow, ensuring that the user hears healthy audio with "reasonable dynamic range, frequency response adapted to the human ear, and safe volume."

[0073] The audio processing system provided in this application embodiment, through the coordinated process of "detection-initialization-dynamic expansion-frequency compensation-conversion output", defines the effective scenarios for subsequent processing by the detection module, provides the parameter basis for the core processor by the initialization module, optimizes the dynamic range of the signal by the expansion module, corrects the frequency response by the compensation module, and ensures accurate signal output by the conversion module. This not only solves the problems of loss of details in weak signals and insufficient perception of high and low frequencies at low volumes, but also avoids transient distortion and gain abrupt changes. The final output is audio with reasonable dynamics, frequency adaptation, and safe volume. While ensuring that details can be heard clearly, it reduces the need for users to increase the volume due to poor listening experience, achieving the dual goals of "natural listening experience + hearing protection".

[0074] Example 2 like Figure 3 The diagram shown is a flowchart of an audio processing method according to an embodiment of this application. The audio processing method provided in this embodiment includes the following steps: Step S110: Real-time detection of the volume of the original digital audio signal; when the volume of the original digital audio signal is less than the first preset sound pressure level threshold, trigger the healthy listening mode. Step S120: After entering the healthy listening mode, read the pre-stored user parameters and initialize the dynamic range extension processor and audio compensation processor based on the user parameters; Step S130: The original digital audio signal is dynamically extended using the dynamic range extension processor to obtain the extended digital audio signal; Step S140: The audio compensation processor is used to perform loudness compensation on the low-frequency and high-frequency bands of the extended digital audio signal to obtain the compensated digital audio signal. Step S150: Convert the compensated digital audio signal into an analog audio signal and play the analog audio signal according to a preset output standard.

[0075] The audio processing method provided in this application embodiment can implement all processes of the audio processing system corresponding to Embodiment 1 and achieve the same technical effect. To avoid repetition, it will not be described again here.

[0076] The audio processing method provided in this application embodiment, through the coordinated process of "detection-initialization-dynamic expansion-frequency compensation-conversion output", defines the effective scenarios for subsequent processing by the detection module, provides the parameter basis for the core processor by the initialization module, optimizes the dynamic range of the signal by the expansion module, corrects the frequency response by the compensation module, and ensures accurate signal output by the conversion module. This not only solves the problems of loss of details in weak signals and insufficient perception of high and low frequencies at low volumes, but also avoids transient distortion and abrupt gain changes. The final output is audio with reasonable dynamics, frequency adaptation, and safe volume. While ensuring that details can be heard clearly, it reduces the need for users to increase the volume due to poor listening experience, achieving the dual goals of "natural listening experience + hearing protection".

[0077] Example 3 This application also provides a computer device. Please refer to the following for details. Figure 4 , Figure 4 This is a basic structural block diagram of the computer device in this embodiment.

[0078] The computer device 4 includes a memory 41, a processor 42, and a network interface 43 that are interconnected via a system bus. It should be noted that only the computer device 4 with memory 41, processor 42, and network interface 43 is shown in the figure; however, it should be understood that it is not required to implement all the components shown, and more or fewer components can be implemented alternatively. Those skilled in the art will understand that the computer device described here is a device capable of automatically performing numerical calculations and / or information processing according to pre-set or stored instructions, and its hardware includes, but is not limited to, microprocessors, application-specific integrated circuits (ASICs), field-programmable gate arrays (FPGAs), digital signal processors (DSPs), embedded devices, etc.

[0079] The computer device can be a desktop computer, laptop, handheld computer, or cloud server, etc. The computer device can interact with the user via a keyboard, mouse, remote control, touchpad, or voice control.

[0080] The memory 41 includes at least one type of readable storage medium, including flash memory, hard disk, multimedia card, card-type memory (e.g., SD or D slot compatibility test memory), random access memory (RAM), static random access memory (SRAM), read-only memory (ROM), electrically erasable programmable read-only memory (EEPROM), programmable read-only memory (PROM), magnetic memory, disk, optical disk, etc. In some embodiments, the memory 41 may be an internal storage unit of the computer device 4, such as the hard disk or memory of the computer device 4. In other embodiments, the memory 41 may also be an external storage device of the computer device 4, such as a plug-in hard disk, smart media card (SMC), secure digital card (SD), flash card, etc., equipped on the computer device 4. Of course, the memory 41 may also include both the internal storage unit and its external storage device of the computer device 4. In this embodiment, the memory 41 is typically used to store the operating system and various application software installed on the computer device 4, such as computer-readable instructions for slot compatibility testing methods. In addition, the memory 41 can also be used to temporarily store various types of data that have been output or will be output.

[0081] In some embodiments, the processor 42 may be a central processing unit (CPU), controller, microcontroller, microprocessor, or other audio processing chip. The processor 42 is typically used to control the overall operation of the computer device 4. In this embodiment, the processor 42 is used to execute computer-readable instructions stored in the memory 41 or to process data, such as executing computer-readable instructions for the slot compatibility testing method.

[0082] The network interface 43 may include a wireless network interface or a wired network interface, which is typically used to establish communication connections between the computer device 4 and other electronic devices.

[0083] The computer device provided in this embodiment can execute the above-described audio processing method. The audio processing method here can be any of the audio processing methods described in the various embodiments above.

[0084] Example 4 This embodiment also provides a computer-readable storage medium storing a computer program thereon, which, when executed by a processor, implements the steps of the audio processing method in this embodiment.

[0085] In this embodiment, the computer-readable storage medium includes flash memory, hard disk, multimedia card, card-type memory (e.g., SD or DX memory), random access memory (RAM), static random access memory (SRAM), read-only memory (ROM), electrically erasable programmable read-only memory (EEPROM), programmable read-only memory (PROM), magnetic memory, magnetic disk, optical disk, etc. In some embodiments, the computer-readable storage medium can be an internal storage unit of a computer device, such as the hard disk or memory of the computer device. In other embodiments, the computer-readable storage medium can also be an external storage device of the computer device, such as a plug-in hard disk, smart media card (SMC), secure digital (SD) card, flash card, etc., equipped on the computer device. Of course, the computer-readable storage medium can also include both the internal storage unit and the external storage device of the computer device. In this embodiment, the computer-readable storage medium is typically used to store the operating system and various application software installed on the computer device. In addition, the computer-readable storage medium can also be used to temporarily store various types of data that have been output or will be output.

[0086] In the several embodiments provided in this application, it should be understood that the disclosed systems and methods can also be implemented in other ways. The system embodiments described above are merely illustrative; for example, the flowcharts and block diagrams in the accompanying drawings illustrate the architecture, functionality, and operation of possible implementations of systems, methods, and computer program products according to various embodiments of the present invention. In this regard, each block in a flowchart or block diagram may represent a module, segment, or portion of code containing one or more executable instructions for implementing a specified logical function. It should also be noted that, as an alternative implementation, the functions marked in the blocks may occur in a different order than those marked in the drawings. For example, two consecutive blocks may actually be executed substantially in parallel, and they may sometimes be executed in reverse order, depending on the functions involved. It should also be noted that each block in the block diagram and / or flowchart, and combinations of blocks in the block diagram and / or flowchart, can be implemented using a dedicated hardware-based system that performs the specified function or action, or using a combination of dedicated hardware and computer instructions.

[0087] In addition, the functional modules or units in the various embodiments of the present invention can be integrated together to form an independent part, or each module can exist independently, or two or more modules can be integrated to form an independent part.

[0088] If the aforementioned functions are implemented as software functional modules and sold or used as independent products, they can be stored in a computer-readable storage medium. Based on this understanding, the technical solution of this invention, or the part that contributes to the prior art, or a part of the technical solution, can be embodied in the form of a software product. This computer software product is stored in a storage medium and includes several instructions to cause a computer device (which may be a smartphone, personal computer, server, or network device, etc.) to execute all or part of the steps of the methods described in the various embodiments of this invention. The aforementioned storage medium can be a non-volatile storage medium or a volatile storage medium. For example, the storage medium can be a USB flash drive, a portable hard drive, a read-only memory (ROM), a random access memory (RAM), a magnetic disk, or an optical disk, or any other medium capable of storing program code.

[0089] The above description is merely a specific embodiment of the present invention, but the scope of protection of the present invention is not limited thereto. Any changes or substitutions that can be easily conceived by those skilled in the art within the scope of the technology disclosed in the present invention should be included within the scope of protection of the present invention.

Claims

1. An audio processing system, characterized in that, The system includes: The detection module is used to detect the volume of the original digital audio signal in real time. When the volume of the original digital audio signal is less than a first preset sound pressure level threshold, the healthy listening mode is triggered. An initialization module is used to read pre-stored user parameters after entering the healthy listening mode, and initialize the dynamic range extension processor and audio compensation processor based on the user parameters; An extension module is used to extend the dynamic range of the original digital audio signal using the dynamic range extension processor to obtain an extended digital audio signal; The compensation module is used to perform loudness compensation on the low-frequency and high-frequency bands of the extended digital audio signal using the audio compensation processor to obtain the compensated digital audio signal. The conversion module is used to convert the compensated digital audio signal into an analog audio signal and play the analog audio signal according to a preset output standard.

2. The audio processing system according to claim 1, characterized in that, The extension module includes: The signal analysis unit is used to calculate the absolute value of the original digital audio signal, calculate the peak value of the original digital audio signal based on the absolute value, and calculate the root mean square value of the original digital audio signal. An adaptive parameter adjustment unit is used to determine the transient factor of the original digital audio signal based on the root mean square value and peak value of the original digital audio signal, and to determine the target spread ratio based on the transient factor.

3. The audio processing system according to claim 2, characterized in that, The adaptive parameter adjustment unit includes: The normalization subunit is used to linearly convert the original digital audio signal whose root mean square value is within a preset sound pressure level range to a preset numerical range to obtain the converted digital audio signal. A transient detection subunit is used to determine whether the converted digital audio signal is a transient signal based on the difference between the peak value and the root mean square value. When the difference is greater than a first preset transient threshold, the converted digital audio signal is identified as a strong transient signal. When the difference is less than or equal to the first preset transient threshold and greater than a second preset transient threshold, the converted digital audio signal is identified as a weak transient signal. When the difference is less than or equal to the second preset transient threshold, the converted digital audio signal is identified as having no transient signal. The unit also determines the transient factor of the converted digital audio signal and determines the target spread ratio based on the transient factor.

4. The audio processing system according to claim 3, characterized in that, The extension module also includes: An envelope detection unit is used to calculate the absolute value of the converted digital audio signal and the current envelope value, and to calculate the target envelope value based on the absolute value of the converted digital audio signal and the current envelope value. The gain calculation unit is used to calculate the target gain using the target spread ratio and the target envelope value, and update the original digital audio signal according to the target gain to obtain the expanded digital audio signal.

5. The audio processing system according to claim 4, characterized in that, The envelope detection unit includes: The mode selection subunit is used to enter attack mode when the absolute value of the converted digital audio signal is greater than the current envelope value, and to enter release mode when the absolute value of the converted digital audio signal is less than the current envelope value. An envelope update subunit is used to calculate the target envelope value of the converted digital audio signal using a preset attack coefficient, the absolute value, and the current envelope value in the attack mode, and to calculate the target envelope value of the converted digital audio signal using a preset release coefficient, the absolute value, and the current envelope value in the release mode.

6. The audio processing system according to claim 4, characterized in that, The gain calculation unit includes: The soft inflection point processing subunit is used to determine the upper half inflection point and the lower half inflection point using a second preset sound pressure level threshold. When the sound pressure level of the converted digital audio signal is greater than the upper half inflection point, the converted digital audio signal is directly used as the digital audio signal after soft inflection point processing. When the sound pressure level of the converted digital audio signal is less than or equal to the upper half inflection point and greater than or equal to the lower half inflection point, the converted digital audio signal is smoothed using a quadratic curve to obtain the digital audio signal after soft inflection point processing. When the sound pressure level of the converted digital audio signal is less than the lower half inflection point, the converted digital audio signal is fixed to a preset gain value to obtain the digital audio signal after soft inflection point processing. The gain calculation subunit is used to calculate the target gain based on the digital audio signal after the soft inflection point processing, and using a preset dynamic compensation threshold, the target spread ratio and the target envelope value. The gain update subunit is used to determine the smoothing coefficients based on the attack mode or release mode of the converted digital audio signal, calculate the update gain based on the target gain and the smoothing coefficients, and update the original digital audio signal using the update gain to obtain the extended digital audio signal.

7. The audio processing system according to claim 1, characterized in that, The compensation module includes: A low-shelf filter is used to perform gain compensation on frequency bands less than a first preset frequency in the extended digital audio signal based on the perceived attenuation characteristics of the low-frequency band in the preset equal loudness curve. A high-shelf filter is used to perform gain compensation on frequency bands above a second preset frequency in the extended digital audio signal based on the perceived attenuation characteristics of the high-frequency band in the preset equal loudness curve.

8. An audio processing method, characterized in that, The method includes: The volume of the original digital audio signal is detected in real time. When the volume of the original digital audio signal is less than a first preset sound pressure level threshold, a healthy listening mode is triggered. After entering the healthy listening mode, the pre-stored user parameters are read, and the dynamic range extension processor and audio compensation processor are initialized based on the user parameters; The original digital audio signal is dynamically extended using the dynamic range extender to obtain an extended digital audio signal. The audio compensation processor is used to perform loudness compensation on the low-frequency and high-frequency bands of the extended digital audio signal to obtain the compensated digital audio signal. The compensated digital audio signal is converted into an analog audio signal, and the analog audio signal is played according to a preset output standard.

9. A computer device, characterized in that, It includes a memory and a processor, the memory storing a computer program, and the processor executing the computer program to implement the steps of the audio processing method of claim 8.

10. A computer-readable storage medium, characterized in that, The computer-readable storage medium stores a computer program that, when executed by a processor, implements the steps of the audio processing method of claim 8.

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