Intelligent regulation and control method and system for loudspeaker audio signals and storage medium

Abstract

The invention discloses an intelligent regulation and control method and system for a loudspeaker audio signal and a storage medium, and relates to the field of audio signal analysis, and the method comprises the steps: monitoring a loudspeaker, and obtaining a pilot run monitoring result of the loudspeaker; collecting an audio signal and environmental noise, and segmenting the enhanced audio signal into short-time frames according to a preset time period; performing comprehensive analysis and processing to generate a definition evaluation value of the audio signal to be played, and performing comprehensive analysis and processing to obtain an audio quality evaluation value of the audio signal to be played; and comprehensively analyzing and processing to obtain an audio signal regulation and control value set of the audio signal to be played, regulating and controlling, and outputting. According to the method, the temperature and the amplitude of the voice coil are monitored through trial operation, the audio signal is segmented into the short-time frames, the processing efficiency and the accuracy are improved, and targeted regulation and control are achieved. Meanwhile, the environmental noise evaluation value is collected, the audio signal is adjusted to adapt to different noise environments, and it is ensured that the output sound quality is clearer.

Description

Technical Field

[0001] The present invention relates to the field of audio signal analysis, and in particular to an intelligent control method, system and storage medium for loudspeaker audio signals. Background Art

[0002] With the popularization of digital audio technology, the processing and transmission of audio signals have become more flexible and complex, requiring more advanced control methods to ensure sound quality. The popularity of smart speakers, smart car systems and other devices has led to higher and higher requirements for audio quality from users, which requires more flexible control methods.

[0003] The prior art, such as the invention patent with announcement number: CN114724580B, is an audio signal control method, the method includes: accessing the original audio signal, inputting the original audio signal into the post-processing module for adjustment, and obtaining the first audio signal; adjusting the gain of the original audio signal to obtain the second audio signal; filtering out the middle and high frequency signals in the second audio signal to obtain the third audio signal; equalizing and adjusting the third audio signal to obtain the fourth audio signal; performing dynamic range control on the fourth audio signal to obtain the fifth audio signal; limiting the fifth audio signal to obtain the sixth audio signal. A speaker system includes a sound source input, a digital signal processor with multi-channel independent control, an amplifier, a cabinet, and a first speaker unit and a second speaker unit respectively receiving two audio signals after amplifying the first audio signal and the sixth audio signal.

[0004] The prior art, such as the invention patent with announcement number: CN101453532B, is a sound processing device for a speaker switch, including a microphone input signal envelope estimator, a microphone input signal background noise envelope estimator, a microphone input signal useful audio envelope estimator, a microphone input signal attenuator, a speaker received signal envelope estimator, a speaker received signal background noise envelope estimator, a speaker received signal useful audio envelope estimator, a speaker received signal attenuator and a state machine for controlling the conversion of the above two states.

[0005] Based on the above scheme, it can be seen that the existing technology in the field of audio signal control usually focuses on processing and analyzing the audio signal itself. However, in actual applications, the input and output of the audio signal will be affected by many factors, such as environmental noise. If only the audio signal is analyzed and processed, the audio signal will usually be distorted or the control parameters will be inaccurate. Summary of the invention

[0006] The present invention aims to solve at least one of the technical problems existing in the prior art. To this end, the present invention provides an intelligent control method, system and storage medium for speaker audio signals, which can divide the audio signals into short time frames, improve processing efficiency and accuracy, and achieve targeted control.

[0007] On the one hand, an embodiment of the present invention provides an intelligent control method for a loudspeaker audio signal, comprising: The speaker to be controlled performs a trial run after receiving the start signal, monitors the real-time performance of the voice coil and the real-time performance of the speaker diaphragm of the speaker to be controlled during the trial run, and obtains the hardware trial run monitoring result of the speaker to be controlled; When the hardware trial operation monitoring result of the speaker to be controlled shows that the performance is normal, a permission formal start instruction is sent, and the audio signal to be played and the environmental noise of the speaker to be controlled are collected at the same time, and the audio signal is preliminarily enhanced in sound quality, and the enhanced audio signal is divided into short time frames according to a preset time period, which are recorded as each frame audio signal; Calculate the frequency domain features of each frame of the audio signal, comprehensively analyze and process to generate a clarity evaluation value of the audio signal to be played, and extract the linear prediction coding coefficient to obtain the formant parameters of each frame of the audio signal, and comprehensively analyze and process to obtain the audio quality evaluation value of the audio signal to be played; Based on the clarity evaluation value and the audio quality evaluation value of the audio signal to be played, a comprehensive analysis and processing is performed to obtain an audio signal control value set of the audio signal to be played, which is then controlled and output.

[0008] According to some embodiments of the present invention, the real-time performance of the voice coil and the real-time performance of the speaker diaphragm to be regulated during the monitoring trial operation is specifically performed as follows: Monitor the real-time performance of the voice coil of the speaker to be regulated during the trial operation, and obtain the temperature performance parameters of the voice coil, including the temperature change rate, maximum temperature, minimum temperature and average temperature of the voice coil during the trial operation period, thereby analyzing and processing to obtain the thermal stability evaluation value of the voice coil of the speaker to be regulated; Monitor the real-time performance of the speaker diaphragm of the speaker to be controlled during the trial operation, and obtain the performance parameters of the speaker diaphragm, including the amplitude change rate, maximum amplitude, minimum amplitude, average amplitude and frequency response average time of the speaker diaphragm during the trial operation period, thereby analyzing and processing to obtain the amplitude stability evaluation value of the speaker diaphragm of the speaker to be controlled.

[0009] According to some embodiments of the present invention, the specific process of obtaining the hardware trial operation monitoring result of the speaker to be regulated is as follows: Based on the thermal stability evaluation value of the voice coil of the speaker to be controlled and the amplitude stability evaluation value of the speaker diaphragm, a comparison analysis is performed with the thermal stability evaluation threshold of the voice coil and the amplitude stability evaluation threshold of the speaker diaphragm pre-stored in the database respectively; when the thermal stability evaluation value of the voice coil of the speaker to be controlled is greater than the thermal stability evaluation threshold and the amplitude stability evaluation value of the speaker diaphragm of the speaker to be controlled is greater than the amplitude stability evaluation threshold, it is determined that the hardware trial operation monitoring result of the speaker to be controlled is normal in performance; If the thermal stability evaluation value of the voice coil of the speaker to be controlled is less than or equal to the thermal stability evaluation threshold, or the amplitude stability evaluation value of the speaker diaphragm of the speaker to be controlled is less than or equal to the amplitude stability evaluation threshold, then the hardware trial operation monitoring result of the speaker to be controlled is determined to be performance abnormality.

[0010] According to some embodiments of the present invention, collecting the audio signal to be played and the ambient noise of the speaker to be adjusted, performing preliminary sound quality enhancement on the audio signal, and dividing the enhanced audio signal into short time frames according to a preset time period specifically includes: After sending a permission formal start instruction to the control terminal of the speaker to be controlled, the audio source to be played is connected to the audio processing unit of the speaker to be controlled to collect audio signals; Based on the externally configured environmental noise sensor of the speaker to be controlled, the environmental noise data of the speaker to be controlled within a preset period is extracted, including the average decibel value of the noise, the equivalent continuous noise level, the peak noise level and the noise spectrum, and the environmental noise evaluation value of the speaker to be controlled is obtained by analysis and processing; The environmental noise evaluation value is used to characterize the degree of influence of the environmental noise on the audio signal to be played; Based on the environmental noise evaluation value of the speaker to be controlled, a mapping and matching is performed with a sound quality enhancement parameter set corresponding to each environmental noise evaluation value interval pre-stored in the database to obtain a sound quality enhancement parameter of the audio signal to be played, and the audio signal is preliminarily enhanced in sound quality based on the sound quality enhancement parameter to obtain an enhanced audio signal; The preset time period size is extracted from the database, and the enhanced audio signal is divided into short time frames, which are recorded as each frame audio signal.

[0011] According to some embodiments of the present invention, the step of calculating the frequency domain features of each frame of the audio signal and generating the clarity evaluation value of the audio signal to be played by comprehensive analysis and processing specifically includes: The frequency domain characteristics of each frame of audio signal are calculated based on fast Fourier transform, including the frequency center, bandwidth and spectral flatness of each frame of audio signal, and the frequency domain characteristics of each frame of audio signal are analyzed and processed to obtain the average bandwidth, frequency center fluctuation rate and average spectral flatness of the audio signal to be played; A set of clarity index parameters is extracted from the database, including bandwidth index, frequency center fluctuation index and spectral flatness index. The average bandwidth, frequency center fluctuation rate and average spectral flatness of the audio signal to be played are compared and analyzed with the clarity index parameter set to obtain a clarity evaluation value of the audio signal to be played.

[0012] According to some embodiments of the present invention, extracting the linear prediction coding coefficients to obtain the formant parameters of each frame of the audio signal, and performing comprehensive analysis and processing to obtain the audio quality evaluation value of the audio signal to be played specifically includes: Calculating the autocorrelation coefficient of each frame of audio signal, and calculating again based on the autocorrelation coefficient of each frame of audio signal to obtain the linear prediction coding coefficient of each frame of audio signal; Based on the linear prediction coding coefficients of each frame of audio signal, the formant parameters of each frame of audio signal are obtained, including the formant position, the formant bandwidth and the formant intensity, and the formant parameters of each frame of audio signal are analyzed and processed to obtain the formant position fluctuation rate, the average formant bandwidth and the formant intensity fluctuation rate of the audio signal to be played; Audio quality index parameters are extracted from a database, including a resonance peak position fluctuation rate index, an average resonance peak bandwidth index and a resonance peak intensity fluctuation rate index. The resonance peak position fluctuation rate, the average resonance peak bandwidth and the resonance peak intensity fluctuation rate of the audio signal to be played are compared and analyzed with the audio quality index parameters to obtain an audio quality evaluation value of the audio signal to be played.

[0013] According to some embodiments of the present invention, the comprehensive analysis process obtains the audio signal control value set of the audio signal to be played and controls it before outputting it. The specific processing conditions are: Based on the clarity evaluation value and the audio quality evaluation value of the audio signal to be played, a comprehensive characteristic value of the audio signal to be played is obtained by comprehensive analysis and processing, and the comprehensive characteristic value of the audio signal to be played is mapped and matched with a set of audio signal control parameters corresponding to each comprehensive characteristic value interval pre-stored in the database to obtain a set of control parameters of the audio signal to be played, including a high-frequency control value, a low-frequency control value and an audio compression value; Based on the control parameter set of the audio signal, the audio signal to be played is advanced controlled.

[0014] According to some embodiments of the present invention, the comprehensive analysis process obtains the comprehensive characteristic value of the audio signal to be played, and the specific processing conditions are: ; in, is the comprehensive characteristic value of the audio signal to be played, is the clarity evaluation value of the audio signal to be played, is the audio quality evaluation value of the audio signal to be played, is the clarity weight factor, is the audio quality weight factor, is the activation function, .

[0015] In a second aspect, an embodiment of the present invention provides an intelligent control system for a loudspeaker audio signal, comprising: A trial operation module is used to perform a trial operation on the speaker to be controlled after receiving a start signal, monitor the real-time performance of the voice coil and the real-time performance of the speaker diaphragm of the speaker to be controlled during the trial operation, and obtain the hardware trial operation monitoring result of the speaker to be controlled; The sound quality preliminary enhancement module is used to send a permission formal start instruction when the hardware trial operation monitoring result of the speaker to be controlled is normal, collect the audio signal to be played and the environmental noise of the speaker to be controlled, perform preliminary sound quality enhancement on the audio signal, and divide the enhanced audio signal into short time frames according to a preset time period, which are recorded as each frame audio signal; The audio feature analysis module is used to calculate the frequency domain features of each frame of the audio signal, comprehensively analyze and process to generate the clarity evaluation value of the audio signal to be played, and extract the linear prediction coding coefficient to obtain the formant parameters of each frame of the audio signal, and comprehensively analyze and process to obtain the audio quality evaluation value of the audio signal to be played; The audio control output module is used to comprehensively analyze and process the clarity evaluation value and audio quality evaluation value of the audio signal to be played, obtain the audio signal control value set of the audio signal to be played, and output it after control.

[0016] In a third aspect, an embodiment of the present invention provides a storage medium, including: the storage medium has one or more programs, and the one or more programs are executed by one or more processors.

[0017] Compared with the prior art, the embodiments of the present invention have at least the following beneficial effects: (1) The present invention provides an intelligent control method for speaker audio signals, which prevents speaker damage due to overheating and mechanical damage due to excessive amplitude by testing the speaker to be controlled at startup and monitoring the real-time performance of the voice coil and speaker diaphragm. At the same time, potential fault signs such as abnormal temperature rise or abnormal amplitude fluctuation can be discovered in advance, and audio signal distortion caused by the speaker's own performance defects can be avoided to a large extent.

[0018] (2) The present invention divides the audio signal into short-time frames and then analyzes and processes each frame of the audio signal, thereby more accurately capturing and analyzing the instantaneous characteristics of the signal, facilitating fine processing, and improving analysis efficiency and accuracy. By comprehensively analyzing the analysis results of each frame of the audio signal, the clarity evaluation value and audio quality evaluation value of the audio signal to be played are calculated, thereby improving the efficiency of audio analysis and processing, being able to perform corresponding regulation and processing on specific audio features, and facilitating faster analysis. At the same time, since the focus is on local features, the accuracy of the analysis is also improved.

[0019] (3) The present invention collects the environmental noise of the speaker to be controlled, analyzes and processes it to obtain an environmental noise evaluation value of the speaker to be controlled. According to the environmental noise evaluation value, the audio signal can be adjusted in a targeted manner to enhance the sound quality so as to adapt to different noise environments and ensure that the sound quality of the audio signal actually outputted is clearer.

[0020] Of course, any product implementing the present invention does not necessarily need to achieve all of the above advantages at the same time. BRIEF DESCRIPTION OF THE DRAWINGS

[0021] Figure 1 It is a schematic diagram of the method flow of the present invention.

[0022] Figure 2 It is a schematic diagram of the system module of the present invention. DETAILED DESCRIPTION

[0023] Embodiments of the present invention are described in detail below, examples of which are shown in the accompanying drawings, wherein the same or similar reference numerals throughout represent the same or similar elements or elements having the same or similar functions. The embodiments described below with reference to the accompanying drawings are exemplary and are only used to explain the present invention, and cannot be understood as limiting the present invention.

[0024] In the description of the present invention, it is necessary to understand that "several" means one or more, "multiple" means more than two, greater than, less than, exceeding, etc. are understood to exclude the number itself, and "above", "below", "within", etc. are understood to include the number itself. If there is a description of "first", "second", etc., it is only for the purpose of distinguishing technical features, and cannot be understood as indicating or implying relative importance or implicitly indicating the number of the indicated technical features or implicitly indicating the order of the indicated technical features. The terms "opening", "upper", "lower", "thickness", "top", "middle", "length", "inside", "around", etc. indicate orientation or positional relationship, which is only for the convenience of describing the present invention and simplifying the description, rather than indicating or implying that the components or elements referred to must have a specific orientation, be constructed and operated in a specific orientation, and therefore cannot be understood as limiting the present invention.

[0025] See also Figure 1 As shown, an embodiment of the present invention provides an intelligent control method for a loudspeaker audio signal, comprising: The speaker to be controlled performs a trial run after receiving the start signal, monitors the real-time performance of the voice coil and the real-time performance of the speaker diaphragm of the speaker to be controlled during the trial run, and obtains the hardware trial run monitoring result of the speaker to be controlled.

[0026] The real-time performance of the voice coil and the real-time performance of the speaker diaphragm to be regulated during the monitoring trial operation is specifically as follows: Monitor the real-time performance of the voice coil of the speaker to be regulated during the trial operation, and obtain the temperature performance parameters of the voice coil, including the temperature change rate, maximum temperature, minimum temperature and average temperature of the voice coil during the trial operation period.

[0027] The temperature change rate refers to the rate of change of the voice coil temperature during the test run period. It reflects the sensitivity and response speed of the voice coil to temperature changes. A higher temperature change rate usually means that the voice coil has experienced drastic temperature fluctuations in a short period of time.

[0028] Maximum temperature refers to the highest temperature value reached by the voice coil during the test run period. It indicates the thermal performance of the voice coil under extreme conditions. Excessive maximum temperature usually leads to aging of the voice coil material, performance degradation or damage.

[0029] The minimum temperature refers to the lowest temperature value reached by the voice coil during the test run period. If the temperature is too low, the material will become brittle or the performance will change.

[0030] Average temperature refers to the average temperature value of the voice coil during the test run period. It provides the overall thermal performance of the voice coil during the test run period. An average temperature that is too high usually indicates insufficient heat dissipation or excessive power, while an average temperature that is too low usually indicates that the speaker is not effectively utilized. When evaluating thermal stability, the average temperature should be kept within a reasonable range to ensure long-term stable operation of the voice coil.

[0031] The maximum temperature change rate of the voice coil is extracted from the database, and the thermal stability evaluation value of the voice coil of the speaker to be controlled is obtained by analysis and processing, specifically including: ; in, is the thermal stability evaluation value of the voice coil of the speaker to be regulated, is the temperature change rate of the voice coil, is the maximum temperature of the voice coil, is the minimum temperature of the voice coil, is the average temperature of the voice coil, It is the maximum temperature change rate that the voice coil can withstand.

[0032] It should also be noted that the temperature change rate, maximum temperature, minimum temperature and average temperature of the voice coil during the test run cycle have certain correlations, which can reflect the thermal stability of the voice coil. The temperature change rate refers to the rate of temperature change within a certain period of time. If the temperature change rate is high, it usually means that the voice coil has experienced a rapid temperature rise or fall, which usually results in a high maximum temperature or a low minimum temperature. A large temperature change rate usually indicates that the speaker has been subjected to a large power change or heat load during operation, which usually affects the thermal stability of the voice coil. The maximum temperature is the highest temperature value reached during the test run cycle, and the average temperature is the average temperature over the entire cycle. If the maximum temperature is much higher than the average temperature, this usually indicates that there is a short-term overheating phenomenon, usually caused by instantaneous high power input or poor heat dissipation. The minimum temperature is the lowest temperature value reached during the test run cycle. If the minimum temperature is much lower than the average temperature, this usually indicates that the speaker has hardly worked for a period of time or the ambient temperature is low. The temperature change rate will also affect the average temperature. If the temperature change rate is large, but the difference between the maximum and minimum temperatures is not large, then the average temperature is usually relatively stable. Conversely, if the temperature change rate is small, but the difference between the maximum and minimum temperatures is large, the average temperature is usually affected more. Together, these parameters reflect the thermal performance of the voice coil.

[0033] Monitor the real-time performance of the speaker diaphragm of the speaker to be regulated during the trial operation, and obtain the performance parameters of the speaker diaphragm, including the amplitude change rate, maximum amplitude, minimum amplitude, average amplitude and average frequency response time of the speaker diaphragm during the trial operation period, and extract the maximum tolerable amplitude change rate and target frequency response average time from the database.

[0034] The amplitude change rate refers to the change in amplitude during the test run cycle. In oscillation, the amplitude refers to the displacement from the equilibrium position to other positions. The amplitude at the last moment of the test run cycle is subtracted from the amplitude at the beginning of the test run cycle to obtain the change in amplitude during the test run cycle. The amplitude change rate reflects the dynamic response capability and stability of the speaker diaphragm. When evaluating stability, a low and stable amplitude change rate is an ideal choice.

[0035] The maximum amplitude refers to the maximum amplitude value recorded by the sensor during the test run period. It is used to indicate the maximum vibration amplitude that the speaker diaphragm can withstand.

[0036] The minimum amplitude refers to the minimum amplitude value of the speaker diaphragm that the sensor can record during the test run period. During the test run period, the test amplitude range covers the rated minimum amplitude value and the rated maximum amplitude value of the speaker diaphragm.

[0037] The average amplitude is the average value of the amplitude during the test run period. It provides an overall level of amplitude. A stable average amplitude indicates that the speaker diaphragm is performing stably under normal operating conditions.

[0038] The average frequency response time refers to the average time it takes for the speaker diaphragm to respond to frequency changes. It reflects the frequency response characteristics of the speaker diaphragm, that is, the adaptability of the speaker diaphragm to different frequency signals. A shorter average frequency response time means that the speaker diaphragm can adapt to frequency changes faster and improve stability.

[0039] The amplitude stability evaluation value of the speaker diaphragm of the speaker to be regulated is obtained by the analysis and processing, specifically including: ; in, is the amplitude stability evaluation value of the speaker diaphragm of the speaker to be regulated, is the amplitude change rate of the loudspeaker diaphragm, is the maximum amplitude of the loudspeaker diaphragm, is the minimum amplitude of the speaker diaphragm, is the average amplitude of the loudspeaker diaphragm, is the average frequency response time of the loudspeaker diaphragm, is the maximum tolerable amplitude change rate, is the target frequency response averaging time.

[0040] It should also be noted that there is a certain correlation between the amplitude change rate, maximum amplitude, minimum amplitude, average amplitude and frequency response average time of the speaker diaphragm during the test run period. A high amplitude change rate means that the speaker diaphragm has experienced a rapid change from the minimum amplitude to the maximum amplitude. A stable amplitude change rate is usually associated with a smaller maximum amplitude and a larger minimum amplitude, indicating that the speaker diaphragm is operating within the normal operating range. The difference between the maximum and minimum amplitudes reflects the amplitude range to which the speaker diaphragm can respond. The average amplitude is the middle value between the maximum and minimum amplitudes, reflecting the overall amplitude level of the speaker diaphragm during the test run period. A stable average amplitude indicates that the speaker diaphragm performs stably under normal operating conditions. A short frequency response average time means that the speaker diaphragm can quickly adapt to frequency changes, which is usually associated with a lower amplitude change rate, indicating that the speaker diaphragm has a good ability to adapt to rapidly changing audio signals. A longer frequency response average time usually means that the speaker diaphragm responds slower to frequency changes. The frequency response average time is also usually related to the maximum and minimum amplitudes, especially when the amplitude response of the speaker diaphragm is different at different frequencies. A fast response time usually means that the speaker diaphragm can maintain good performance at different amplitudes.

[0041] The specific process of obtaining the hardware trial operation monitoring result of the speaker to be controlled is as follows: Based on the thermal stability evaluation value of the voice coil of the speaker to be controlled and the amplitude stability evaluation value of the speaker diaphragm, a comparison and analysis are performed with the thermal stability evaluation threshold of the voice coil and the amplitude stability evaluation threshold of the speaker diaphragm pre-stored in the database respectively; when the thermal stability evaluation value of the voice coil of the speaker to be controlled is greater than the thermal stability evaluation threshold and the amplitude stability evaluation value of the speaker diaphragm of the speaker to be controlled is greater than the amplitude stability evaluation threshold, the hardware trial operation monitoring result of the speaker to be controlled is judged to be normal in performance.

[0042] If the thermal stability evaluation value of the voice coil of the speaker to be controlled is less than or equal to the thermal stability evaluation threshold, or the amplitude stability evaluation value of the speaker diaphragm of the speaker to be controlled is less than or equal to the amplitude stability evaluation threshold, then the hardware trial operation monitoring result of the speaker to be controlled is determined to be performance abnormality.

[0043] The thermal stability evaluation threshold is processed as follows: in a specific embodiment, the thermal stability evaluation threshold is set directly in the database during the development of the intelligent control system for the speaker audio signal involved in the embodiment of the present invention. There are many methods for setting the thermal stability evaluation threshold, such as obtaining it through statistical analysis, including counting the thermal stability evaluation values ​​under normal historical thermal stability evaluations (for example, the average temperature of the voice coil is less than 40 degrees Celsius), and averaging the thermal stability evaluation values ​​obtained multiple times to obtain the thermal stability evaluation threshold.

[0044] The amplitude stability evaluation threshold is processed as follows: in a specific embodiment, the amplitude stability evaluation threshold is set directly in the database during the development of the intelligent control system for the speaker audio signal involved in the embodiment of the present invention. There are many methods for setting the amplitude stability evaluation threshold, such as obtaining it through statistical analysis, including counting the amplitude stability evaluation values ​​under normal historical amplitude stability evaluations (for example, the average amplitude of the speaker diaphragm is less than 1mm), and averaging the amplitude stability evaluation values ​​obtained multiple times to obtain the amplitude stability evaluation threshold.

[0045] When the hardware trial operation monitoring result of the speaker to be controlled shows that the performance is normal, a permission formal start instruction is sent, and the audio signal to be played and the ambient noise of the speaker to be controlled are collected at the same time, the audio signal is preliminarily denoised, and the enhanced audio signal is divided into short time frames according to the preset time period, which are recorded as each frame of audio signal.

[0046] The collecting of the audio signal to be played and the ambient noise of the speaker to be adjusted, performing preliminary denoising on the audio signal, and dividing the enhanced audio signal into short time frames according to a preset time period specifically includes: After sending a permission formal start instruction to the control terminal of the speaker to be controlled, the audio source to be played is connected to the audio processing unit of the speaker to be controlled to collect audio signals.

[0047] Based on the externally configured environmental noise sensor of the speaker to be controlled, the environmental noise data of the speaker to be controlled within a preset period is extracted, including the average decibel value of the noise, the equivalent continuous noise level, the peak noise level and the noise spectrum.

[0048] The average noise decibel value refers to the average value of the noise intensity within a preset period, usually in decibels (dB). It is used to evaluate the average noise level around the speaker to be controlled within the preset period.

[0049] The equivalent continuous noise level is to convert each measured noise in a preset period into the corresponding sound power, and calculate the average value of the sound power in the preset period, also in decibels (dB). It is usually used to indicate the average noise level in a preset period. The equivalent continuous noise level is a common indicator for evaluating the impact of noise on people and the environment.

[0050] It should be noted that the step of converting each noise measured within a preset period into a corresponding sound power specifically includes: Convert the sound pressure level to sound pressure. The relationship between the sound pressure level Lp and the sound pressure p is: ; Where p is the sound pressure, is the reference sound pressure, usually 20 micropascals, and Lp is the sound pressure level.

[0051] The sound power is proportional to the square of the sound pressure. The sound power P can be calculated by the following formula: ; in, is the sound power, p is the sound pressure, is the acoustic impedance of the medium. For air, the acoustic impedance is 420 Pascal·s / m (at standard temperature and pressure).

[0052] For each measured noise level, the corresponding sound power is calculated according to the above steps, and then these sound power values ​​are averaged within a preset period to obtain the equivalent sound power.

[0053] Peak noise level refers to the maximum sound pressure level reached by noise at a certain moment within a preset period, measured in decibels (dB). Peak noise level is used to indicate the maximum instantaneous intensity of noise and is very important for evaluating the suddenness and impact of noise.

[0054] Noise spectrum refers to the distribution of noise at different frequencies, usually represented by a graph with frequency as the horizontal axis and sound pressure level as the vertical axis. Noise spectrum is used to analyze the frequency components of noise.

[0055] The environmental noise evaluation value of the speaker to be controlled is obtained by analysis and processing, specifically including: Based on the noise spectrum, the frequency bandwidth and spectrum slope of the environmental noise are obtained through analysis and processing.

[0056] Bandwidth refers to the frequency range of noise energy distribution, that is, the interval between the lowest frequency and the highest frequency covered by the noise. It is usually expressed in Hertz (Hz).

[0057] Spectral slope refers to the rate at which the sound pressure level changes with frequency in a noise spectrum graph, usually expressed in dB / octave or dB / decade. Spectral slope describes the attenuation or growth trend of noise at different frequencies. A positive slope means that the sound pressure level increases as the frequency increases. A negative slope means that the sound pressure level decreases as the frequency increases.

[0058] The noise average decibel indicator, equivalent continuous noise level indicator, peak noise level indicator, frequency band indicator width and spectrum indicator slope are extracted from the database, and the environmental noise evaluation value of the speaker to be controlled is obtained through joint analysis and processing. The specific processing conditions are as follows: ; in, is the environmental noise evaluation value of the speaker to be controlled, is the average decibel value of the ambient noise, is the equivalent continuous noise level of the ambient noise, is the peak noise level of the ambient noise, is the frequency bandwidth of the ambient noise, is the spectrum slope of the ambient noise, is the noise average decibel indicator value, is the equivalent continuous noise level indication value, is the peak noise level indicator, is the frequency band indication width, is the spectrum indicator slope, is the noise average decibel value weight factor, is the equivalent continuous noise level weight factor, is the peak noise level weighting factor, is the bandwidth weight factor, is the spectral slope weight factor. The softplus function is a built-in function in Python. .

[0059] It should be noted that the noise average decibel value weight factor, equivalent continuous noise level weight factor, peak noise level weight factor, bandwidth weight factor and spectrum slope weight factor all have a value range between 0 and 1 and meet The noise average decibel value weight factor is an influence factor corresponding to the environmental noise evaluation value of the speaker to be controlled preset in the database, which indicates the numerical value of the influence of the noise average decibel value on the environmental noise evaluation value of the speaker to be controlled. The equivalent continuous noise level weight factor is an influence factor corresponding to the environmental noise evaluation value of the speaker to be controlled preset in the database, which indicates the numerical value of the influence of the equivalent continuous noise level on the environmental noise evaluation value of the speaker to be controlled. The peak noise level weight factor is an influence factor corresponding to the environmental noise evaluation value of the speaker to be controlled preset in the database, which indicates the numerical value of the influence of the peak noise level on the environmental noise evaluation value of the speaker to be controlled. The bandwidth weight factor is an influence factor corresponding to the environmental noise evaluation value of the speaker to be controlled preset in the database, which indicates the numerical value of the influence of the bandwidth on the environmental noise evaluation value of the speaker to be controlled. The spectral slope weight factor is an influence factor corresponding to the environmental noise evaluation value of the speaker to be controlled preset in the database, and represents a numerical value of the influence degree of the spectral slope on the environmental noise evaluation value of the speaker to be controlled. When used, the noise average decibel value weight factor, equivalent continuous noise level weight factor, peak noise level weight factor, bandwidth weight factor and spectral slope weight factor can be directly obtained from the database. The acquisition method is a pre-set mapping relationship. For example, the noise average decibel value, equivalent continuous noise level, peak noise level, bandwidth and spectral slope involved in the embodiment of the present invention are input into the mapping relationship for mapping and matching to obtain the noise average decibel value weight factor, equivalent continuous noise level weight factor, peak noise level weight factor, bandwidth weight factor and spectral slope weight factor involved in the embodiment of the present invention, and the mapping relationship is one-to-one correspondence.

[0060] It should also be noted that there is a certain correlation between the parameters of noise average decibel value, equivalent continuous noise level, peak noise level, bandwidth and spectrum slope. The noise average decibel value reflects the overall intensity level of noise. Equivalent continuous noise level (Leq) is also an indicator of the overall intensity of noise, but it takes into account the time distribution characteristics of noise. Peak noise level reflects the maximum instantaneous intensity of noise. Bandwidth refers to the frequency range contained in the noise. The wider the bandwidth, the more frequency components the noise contains. The spectrum slope describes the degree of inclination of the noise spectrum. Both the noise average decibel value and the equivalent continuous noise level are indicators of the overall intensity of noise and have a certain positive correlation. A high peak noise level does not mean that the average decibel value or equivalent continuous noise level is also high, because the peak value reflects the instantaneous maximum value, while the average or equivalent value reflects the overall level. The wider the bandwidth, the noise usually contains more high-intensity frequency components, which leads to an increase in the average decibel value or equivalent continuous noise level. The spectrum slope affects the frequency distribution of noise. For example, a steeper slope usually means a stronger low-frequency component, which usually affects the average decibel value of the noise or the equivalent continuous noise level. Noise with a wide bandwidth usually contains more high-intensity frequency components, which usually results in a higher peak noise level. The spectral slope affects the frequency component distribution of the noise, which in turn affects the peak noise level. For example, if the high-frequency component is stronger, the peak noise level will be higher.

[0061] The environmental noise evaluation value is used to characterize the degree of influence of the environmental noise on the audio signal to be played.

[0062] Based on the environmental noise evaluation value of the speaker to be controlled, a mapping and matching is performed with the sound quality enhancement parameter set corresponding to each environmental noise evaluation value interval pre-stored in the database to obtain the sound quality enhancement parameters of the audio signal to be played, and the audio signal is preliminarily enhanced in sound quality based on the sound quality enhancement parameters to obtain an enhanced audio signal.

[0063] The sound quality enhancement parameters include a volume enhancement parameter, a dynamic range compression parameter and a reverberation enhancement parameter.

[0064] Volume boost parameter, controls the amplification of the input signal. Increasing the gain can increase the volume.

[0065] Dynamic range compression parameter that controls the ratio between input signal and output signal. A higher compression ratio will result in a smaller dynamic range and a more balanced sound.

[0066] Reverb enhancement parameter, controls the duration of the reverb effect. A longer reverb time will make the sound fuller, and a shorter reverb time will make the sound clearer.

[0067] For example, the sound quality enhancement parameter set corresponding to the environmental noise evaluation value of a certain speaker to be controlled is: Volume enhancement parameter: +3dB; Dynamic range compression parameters: compression ratio is 2:1, threshold is -80dB; Reverberation enhancement parameters: Reverberation time is 0.5 seconds; Based on the sound quality enhancement parameters, the specific process is as follows: Boost the overall volume of the audio signal by 3dB to offset the effect of ambient noise. Apply a 2:1 compression ratio and a -80dB threshold to reduce the dynamic range of the audio signal, ensuring that weaker sounds can be heard in ambient noise. Add 0.5 seconds of reverberation time to make the audio signal sound fuller and more natural.

[0068] The sound quality enhancement tool involved in the embodiment of the present invention is Audacity.

[0069] The preset time period size is extracted from the database, and the enhanced audio signal is divided into short time frames, which are recorded as each frame audio signal.

[0070] The frequency domain characteristics of each frame of audio signal are calculated, and the clarity evaluation value of the audio signal to be played is generated through comprehensive analysis and processing. At the same time, the linear prediction coding coefficients are extracted to obtain the resonance peak parameters of each frame of audio signal, and the audio quality evaluation value of the audio signal to be played is obtained through comprehensive analysis and processing.

[0071] The step of calculating the frequency domain features of each frame of the audio signal and comprehensively analyzing and processing to generate a clarity evaluation value of the audio signal to be played specifically includes: The frequency domain characteristics of each frame of audio signal are obtained based on fast Fourier transform calculation, including the frequency center, bandwidth and spectral flatness of each frame of audio signal. The frequency domain characteristics of each frame of audio signal are analyzed and processed comprehensively to obtain the average bandwidth, frequency center fluctuation rate and average spectral flatness of the audio signal to be played.

[0072] The Fast Fourier Transform (FFT) is an efficient algorithm for calculating the Discrete Fourier Transform (DFT). DFT is a mathematical transformation that converts discrete time domain signals into discrete frequency domain signals, and FFT provides a method for quickly calculating DFT. FFT takes advantage of the symmetry and periodicity of DFT and reduces the amount of calculation by decomposing the DFT of a long sequence into the DFT of a short sequence. Common FFT algorithms include the Cooley-Tukey algorithm, which calculates the DFT by recursively splitting the sequence into two halves.

[0073] The frequency center is the energy center of the audio signal spectrum, indicating the average frequency of the signal energy. It reflects the pitch of the audio signal, and a higher frequency center usually corresponds to a sharper tone.

[0074] Bandwidth refers to the main frequency range of energy distribution in an audio signal. The larger the bandwidth, the richer the frequency components contained in the signal, and the more complex the sound quality is.

[0075] Spectral flatness is a measure of how evenly energy is distributed across the frequency spectrum of an audio signal. A higher spectral flatness indicates that energy is evenly distributed across frequencies, typically associated with noise. A lower spectral flatness indicates that energy is concentrated at certain frequencies, typically associated with pure tones or specific pitches.

[0076] Average bandwidth refers to the average bandwidth of each frame of audio signal, reflecting the average frequency range of the entire audio signal.

[0077] Frequency center volatility refers to the degree of fluctuation of the frequency center in the time series. A large volatility usually indicates that the pitch of the audio signal varies greatly, which usually affects the stability of auditory perception. The power spectral density (PSD) is calculated to determine the frequency center of the signal at each time point. The frequency center is the frequency corresponding to the maximum value of the power spectral density. The standard deviation of the time-varying sequence of the frequency center is calculated. The larger the standard deviation, the greater the degree of fluctuation of the frequency center. For example, suppose a certain time series data records the frequency center once per second for 100 seconds. The standard deviation of 100 frequency center values ​​is calculated, and this standard deviation is the frequency center volatility.

[0078] Average spectrum flatness refers to the average value of the spectrum flatness of each frame of audio signal, which reflects the average distribution of the spectrum energy of the entire audio signal.

[0079] Extract a set of clarity index parameters from the database, including a bandwidth index, a frequency center fluctuation rate index, and a spectrum flatness index, and compare and analyze the average bandwidth, frequency center fluctuation rate, and average spectrum flatness of the audio signal to be played with the clarity index parameter set to obtain a clarity evaluation value of the audio signal to be played, specifically including: ; in, is the clarity evaluation value of the audio signal to be played, is the average bandwidth of the audio signal to be played, is the frequency center fluctuation rate of the audio signal to be played, is the average spectral flatness of the audio signal to be played, is the bandwidth indicator, is the frequency center volatility indicator, is the spectral flatness index, is the average bandwidth weight factor, is the frequency center volatility weight factor, is the average spectrum flatness weight factor.

[0080] It should be noted that the average bandwidth weight factor, the frequency center volatility weight factor and the average spectrum flatness weight factor all have a value range between 0 and 1 and satisfy The average bandwidth weight factor is an influence factor corresponding to the clarity evaluation value of the audio signal to be played preset in the database, indicating the numerical value of the influence degree of the average bandwidth on the clarity evaluation value of the audio signal to be played. The frequency center fluctuation weight factor is an influence factor corresponding to the clarity evaluation value of the audio signal to be played preset in the database, indicating the numerical value of the influence degree of the frequency center fluctuation on the clarity evaluation value of the audio signal to be played. The average spectrum flatness weight factor is an influence factor corresponding to the clarity evaluation value of the audio signal to be played preset in the database, indicating the numerical value of the influence degree of the average spectrum flatness on the clarity evaluation value of the audio signal to be played. When used, the average bandwidth weight factor, the frequency center fluctuation weight factor and the average spectrum flatness weight factor can be directly obtained from the database, and the acquisition method is a preset mapping relationship. For example, the frequency center fluctuation rate, the average spectrum flatness and the average bandwidth involved in the embodiment of the present invention are input into the mapping relationship for mapping and matching to obtain the average bandwidth weight factor, the frequency center fluctuation weight factor and the average spectrum flatness weight factor involved in the embodiment of the present invention, wherein the mapping relationship is one-to-one correspondence.

[0081] It should also be noted that there is a certain correlation between the average bandwidth, frequency center fluctuation rate and average spectrum flatness of the audio signal to be played. Audio signals with a larger average bandwidth usually have more frequency components, which usually leads to larger fluctuations in the frequency center in time, thereby increasing the frequency center fluctuation rate. Conversely, if the frequency center fluctuation rate is large, it usually means that the audio signal contains frequency components of different ranges in different time frames, which affects the calculation of the average bandwidth. Audio signals with a larger average bandwidth usually contain a wider range of frequency components, which usually leads to higher spectrum flatness because the energy is distributed in a wider frequency range. If the average spectrum flatness is high, it means that the energy distribution is more uniform, which usually means that the signal contains a wider frequency range, thus affecting the average bandwidth. Audio signals with a large frequency center fluctuation rate usually indicate that the tone or main frequency component varies greatly in time, which usually affects the calculation of spectrum flatness because the uniformity of energy distribution usually changes with the change of the frequency center. These three parameters affect each other and jointly determine the frequency domain characteristics of the audio signal.

[0082] The extracting of linear prediction coding coefficients to obtain the formant parameters of each frame of the audio signal and the comprehensive analysis and processing to obtain the audio quality evaluation value of the audio signal to be played specifically include: The autocorrelation coefficient of each frame of the audio signal is calculated, and the linear prediction coding (LPC) coefficient of each frame of the audio signal is calculated again based on the autocorrelation coefficient of each frame of the audio signal.

[0083] The autocorrelation coefficient of each frame of audio signal is a statistic that measures the similarity between each frame of audio signal and itself at different time delays. The autocorrelation coefficient can reveal the periodicity, stationarity and noise characteristics of the signal. In audio signal processing, the autocorrelation coefficient is used to identify the fundamental frequency (i.e., pitch) of the audio. At the same time, the autocorrelation coefficient is the basis for calculating the linear predictive coding (LPC) coefficient.

[0084] The linear predictive coding (LPC) coefficient of each frame of the audio signal is a parameter used to simulate the audio generation mechanism. The LPC coefficient can effectively represent the main characteristics of the audio signal and is used for audio compression and encoding. It is also used to analyze the spectral structure and resonance characteristics of the audio signal.

[0085] Based on the linear predictive coding (LPC) coefficients of each frame of audio signal, the LPC filter is used to obtain the formant parameters of each frame of audio signal, including the formant position sequence, formant bandwidth and formant intensity. The formant parameters of each frame of audio signal are analyzed and processed to obtain the formant position fluctuation rate, average formant bandwidth and formant intensity fluctuation rate of the audio signal to be played.

[0086] The formant position sequence refers to the frequency points where the energy in the audio signal spectrum is concentrated, usually corresponding to the resonant frequency of the vocal tract. The formant position determines the sound quality and timbre of the audio, and the formant position sequence of different vowels is different.

[0087] Resonant peak bandwidth refers to the frequency range of the resonance peak, that is, the width of the resonance peak. Resonant peak bandwidth is related to the damping characteristics of the vocal tract and affects the clarity and naturalness of the sound.

[0088] Formant intensity refers to the amplitude of the formant, which reflects the energy of the frequency component. Formant intensity affects the loudness and perceived volume of the sound.

[0089] Formant position fluctuation refers to the degree to which the sequence of formant positions in an audio signal changes over time. A small fluctuation indicates a stable sound, while a large fluctuation usually indicates an unstable sound or interference. The standard deviation of each formant position sequence is calculated as the formant position fluctuation.

[0090] The average formant bandwidth is the average of all formant bandwidths in the entire audio signal. The average formant bandwidth can be used to assess the overall clarity and naturalness of the sound.

[0091] Formant intensity fluctuation refers to the degree to which the intensity of the formant in the audio signal changes over time. A small intensity fluctuation indicates a stable sound, while a large fluctuation usually indicates a large dynamic range or the presence of noise. The standard deviation of each formant intensity is calculated as the formant intensity fluctuation.

[0092] The position of the formant and the bandwidth directly affect the clarity of the audio. A stable formant position and a moderate bandwidth usually mean higher clarity.

[0093] The fluctuation rate of the position and intensity of the formant can reflect the stability of the sound. A small fluctuation rate indicates a stable sound, while a large fluctuation rate usually indicates an unstable sound or interference.

[0094] Extract audio quality index parameters from the database, including a resonance peak position fluctuation rate index, an average resonance peak bandwidth index, and a resonance peak intensity fluctuation rate index, and compare and analyze the resonance peak position fluctuation rate, the average resonance peak bandwidth, and the resonance peak intensity fluctuation rate of the audio signal to be played with the audio quality index parameters to obtain an audio quality evaluation value of the audio signal to be played, specifically including: ; in, is the audio quality evaluation value of the audio signal to be played, is the fluctuation rate of the formant position of the audio signal to be played, is the average formant bandwidth of the audio signal to be played, is the fluctuation rate of the formant intensity of the audio signal to be played, is the resonance peak position volatility index, is the average resonance peak bandwidth index, is the resonance peak intensity volatility index, is the resonance peak position volatility weight factor, is the average resonance peak bandwidth weight factor, is the resonance peak intensity fluctuation weight factor.

[0095] It should be noted that the value range of the resonance peak position volatility weight factor, the average resonance peak bandwidth weight factor and the resonance peak intensity volatility weight factor are all between 0 and 1 and meet The resonance peak position fluctuation weight factor is an influence factor corresponding to the audio quality evaluation value of the audio signal to be played preset in the database, which indicates the numerical value of the influence degree of the resonance peak position fluctuation on the audio quality evaluation value of the audio signal to be played. The average resonance peak bandwidth weight factor is an influence factor corresponding to the audio quality evaluation value of the audio signal to be played preset in the database, which indicates the numerical value of the influence degree of the average resonance peak bandwidth on the audio quality evaluation value of the audio signal to be played. The resonance peak intensity fluctuation weight factor is an influence factor corresponding to the audio quality evaluation value of the audio signal to be played preset in the database, which indicates the resonance peak intensity fluctuation. The numerical value of the degree of influence on the audio quality evaluation value of the audio signal to be played can be used to directly obtain the resonance peak position fluctuation weight factor, the average resonance peak bandwidth weight factor and the resonance peak intensity fluctuation weight factor from the database, and the acquisition method is a pre-set mapping relationship. For example, the resonance peak position fluctuation rate, the average resonance peak bandwidth and the resonance peak intensity fluctuation rate involved in the embodiment of the present invention are input into the mapping relationship for mapping and matching to obtain the resonance peak position fluctuation rate weight factor, the average resonance peak bandwidth weight factor and the resonance peak intensity fluctuation rate weight factor involved in the embodiment of the present invention, wherein the mapping relationship is one-to-one correspondence.

[0096] It should also be noted that there is a certain correlation between the parameters of the formant position fluctuation rate, the average formant bandwidth and the formant intensity fluctuation rate of the audio signal to be played. A large formant position fluctuation rate usually means that the shape of the vocal tract or the pronunciation method is changing rapidly, which usually causes the formant bandwidth to change. A large formant position fluctuation rate is usually accompanied by a change in bandwidth, especially in a fast speech flow or noisy environment. A large formant position fluctuation rate usually means that the dynamic range of the audio is large, which usually causes the formant intensity to fluctuate. Both reflect the stability and consistency of the audio to a certain extent. Rapid audio changes or noise interference usually cause the formant position and intensity to fluctuate simultaneously. A large average formant bandwidth usually means that the vocal tract attenuates the frequency components faster, which usually causes the formant intensity to fluctuate in time. A large average formant bandwidth usually makes the formant intensity more susceptible to environmental noise or vocal tract changes, resulting in intensity fluctuations. These three parameters together reflect the stability and consistency of the audio. Stable audio usually has a small formant position fluctuation rate, a moderate average formant bandwidth and a small formant intensity fluctuation rate.

[0097] Based on the clarity evaluation value and the audio quality evaluation value of the audio signal to be played, a comprehensive analysis and processing is performed to obtain an audio signal control value set of the audio signal to be played, which is then controlled and output.

[0098] The comprehensive analysis and processing obtains the audio signal control value set of the audio signal to be played and outputs it after control. The specific processing conditions are: Based on the clarity evaluation value and the audio quality evaluation value of the audio signal to be played, a comprehensive characteristic value of the audio signal to be played is obtained by comprehensive analysis and processing, specifically including: ; in, is the comprehensive characteristic value of the audio signal to be played, is the clarity evaluation value of the audio signal to be played, is the audio quality evaluation value of the audio signal to be played, is the clarity weight factor, is the audio quality weight factor, is the activation function, .

[0099] It should be noted that the clarity weight factor and the audio quality weight factor have a value range between 0 and 1 and satisfy The clarity weight factor is an influence factor of the comprehensive characteristic value of the audio signal to be played preset in the database, and represents a numerical value of the influence degree of the clarity evaluation value on the comprehensive characteristic value of the audio signal to be played. The audio quality weight factor is an influence factor of the comprehensive characteristic value of the audio signal to be played preset in the database, and represents a numerical value of the influence degree of the audio quality evaluation value on the comprehensive characteristic value of the audio signal to be played. When used, the clarity weight factor and the audio quality weight factor can be directly obtained from the database, and the acquisition method is a pre-set mapping relationship. For example: the clarity evaluation value and the audio quality evaluation value calculated by the embodiment of the present invention are input into the mapping relationship for mapping and matching to obtain the clarity weight factor and the audio quality weight factor involved in the embodiment of the present invention, and the mapping relationship is one-to-one correspondence.

[0100] The comprehensive characteristic value of the audio signal to be played is mapped and matched with the audio signal control parameter set corresponding to each comprehensive characteristic value interval pre-stored in the database to obtain the control parameter set of the audio signal to be played, including high-frequency control value, low-frequency control value and audio compression value.

[0101] The high frequency control value refers to the parameter for gaining or attenuating the high frequency components (usually frequencies above 2kHz) in the audio signal. Increasing the high frequency gain can improve the clarity and detail of the audio and make the sound brighter.

[0102] The low-frequency control value refers to the parameter for gaining or attenuating the low-frequency components (usually frequencies below 200Hz) in the audio signal. Increasing the low-frequency gain can make the audio sound fuller and more powerful, and enhance the bass effect.

[0103] The audio compression value refers to the parameter for compressing the dynamic range of the audio signal. Compressing the dynamic range of the audio can make the volume more stable and avoid excessive volume fluctuations. Through compression, the overall volume of the audio can be increased, making weaker sounds more obvious.

[0104] For example, an audio signal to be played is collected, and the control parameter set corresponding to the interval where the comprehensive characteristic value of the audio signal is located is: high-frequency control value +3dB, low-frequency control value +2dB, and audio compression value -1dB.

[0105] Apply the acquired control parameters to the audio signal to make corresponding sound quality adjustments, including: Increase the high frequency by 3dB to improve the clarity of the audio, increase the low frequency by 2dB to enhance the fullness of the audio, and reduce the audio compression value by 1dB to reduce distortion.

[0106] Based on the audio signal control parameter set, the audio signal to be played is advanced controlled.

[0107] In this embodiment, the present invention provides an intelligent control system for loudspeaker audio signals, specifically comprising: The trial operation module is used to perform a trial operation on the speaker to be controlled after receiving a start signal, monitor the real-time performance of the voice coil and the real-time performance of the speaker diaphragm of the speaker to be controlled during the trial operation, and obtain the hardware trial operation monitoring result of the speaker to be controlled.

[0108] The module for preliminary sound quality enhancement is used to send a formal start-up instruction when the hardware trial operation monitoring result of the speaker to be controlled shows that the performance is normal. At the same time, it collects the audio signal to be played and the ambient noise of the speaker to be controlled, performs preliminary sound quality enhancement on the audio signal, and divides the enhanced audio signal into short time frames according to the preset time period, which are recorded as each frame of audio signal.

[0109] The audio feature analysis module is used to calculate the frequency domain features of each frame of audio signal, and to generate a clarity evaluation value of the audio signal to be played through comprehensive analysis and processing. At the same time, the linear prediction coding coefficient is extracted to obtain the resonance peak parameters of each frame of audio signal, and the audio quality evaluation value of the audio signal to be played is obtained through comprehensive analysis and processing.

[0110] The audio control output module is used to comprehensively analyze and process the clarity evaluation value and audio quality evaluation value of the audio signal to be played, obtain the audio signal control value set of the audio signal to be played, and output it after control.

[0111] A storage medium includes: the storage medium has one or more programs, and the one or more programs are executed by one or more processors to implement the method as described above.

[0112] It should be noted that, in this article, relational terms such as first and second, etc. are only used to distinguish one entity or operation from another entity or operation, and do not necessarily require or imply any such actual relationship or order between these entities or operations. Moreover, the terms "include", "comprise" or any other variants thereof are intended to cover non-exclusive inclusion, so that a process, method, article or device including a series of elements includes not only those elements, but also other elements not explicitly listed, or also includes elements inherent to such process, method, article or device.

[0113] The preferred embodiments of the present invention disclosed above are only used to help explain the present invention. The preferred embodiments do not describe all the details in detail, nor do they limit the invention to the specific implementation methods described. Obviously, many modifications and changes can be made according to the content of this specification. This specification selects and specifically describes these embodiments in order to better explain the principles and practical applications of the present invention, so that technicians in the relevant technical field can understand and use the present invention well. As long as they do not deviate from the structure of the present invention or exceed the scope defined by the present invention, they should all belong to the protection scope of the present invention.

Claims

1. An intelligent control method for a loudspeaker audio signal, characterized in that: include: The speaker to be controlled performs a trial run after receiving the start signal, monitors the real-time performance of the voice coil and the real-time performance of the speaker diaphragm of the speaker to be controlled during the trial run, and obtains the hardware trial run monitoring result of the speaker to be controlled; When the hardware trial operation monitoring result of the speaker to be controlled shows that the performance is normal, a permission formal start instruction is sent, and the audio signal to be played and the environmental noise of the speaker to be controlled are collected at the same time, and the audio signal is preliminarily enhanced in sound quality, and the enhanced audio signal is divided into short time frames according to a preset time period, which are recorded as each frame audio signal; Calculate the frequency domain features of each frame of the audio signal, comprehensively analyze and process to generate a clarity evaluation value of the audio signal to be played, and extract the linear prediction coding coefficient to obtain the formant parameters of each frame of the audio signal, and comprehensively analyze and process to obtain the audio quality evaluation value of the audio signal to be played; Based on the clarity evaluation value and the audio quality evaluation value of the audio signal to be played, a comprehensive analysis and processing is performed to obtain an audio signal control value set of the audio signal to be played, which is then controlled and output.

2. The intelligent control method of a loudspeaker audio signal according to claim 1, characterized in that: The real-time performance of the voice coil and the real-time performance of the speaker diaphragm to be regulated during the monitoring trial operation is specifically as follows: Monitor the real-time performance of the voice coil of the speaker to be regulated during the trial operation, and obtain the temperature performance parameters of the voice coil, including the temperature change rate, maximum temperature, minimum temperature and average temperature of the voice coil during the trial operation period, thereby analyzing and processing to obtain the thermal stability evaluation value of the voice coil of the speaker to be regulated; Monitor the real-time performance of the speaker diaphragm of the speaker to be controlled during the trial operation, and obtain the performance parameters of the speaker diaphragm, including the amplitude change rate, maximum amplitude, minimum amplitude, average amplitude and frequency response average time of the speaker diaphragm during the trial operation period, thereby analyzing and processing to obtain the amplitude stability evaluation value of the speaker diaphragm of the speaker to be controlled.

3. The intelligent control method of a loudspeaker audio signal according to claim 2, characterized in that: The specific process of obtaining the hardware trial operation monitoring result of the speaker to be controlled is as follows: Based on the thermal stability evaluation value of the voice coil of the speaker to be controlled and the amplitude stability evaluation value of the speaker diaphragm, a comparison analysis is performed with the thermal stability evaluation threshold of the voice coil and the amplitude stability evaluation threshold of the speaker diaphragm pre-stored in the database respectively; when the thermal stability evaluation value of the voice coil of the speaker to be controlled is greater than the thermal stability evaluation threshold and the amplitude stability evaluation value of the speaker diaphragm of the speaker to be controlled is greater than the amplitude stability evaluation threshold, it is determined that the hardware trial operation monitoring result of the speaker to be controlled is normal in performance; If the thermal stability evaluation value of the voice coil of the speaker to be controlled is less than or equal to the thermal stability evaluation threshold, or the amplitude stability evaluation value of the speaker diaphragm of the speaker to be controlled is less than or equal to the amplitude stability evaluation threshold, then the hardware trial operation monitoring result of the speaker to be controlled is determined to be performance abnormality.

4. The intelligent control method of a loudspeaker audio signal according to claim 1, characterized in that: The collecting of the audio signal to be played and the ambient noise of the speaker to be adjusted, performing preliminary sound quality enhancement on the audio signal, and dividing the enhanced audio signal into short time frames according to a preset time period specifically includes: After sending a permission formal start instruction to the control terminal of the speaker to be controlled, the audio source to be played is connected to the audio processing unit of the speaker to be controlled to collect audio signals; Based on the externally configured environmental noise sensor of the speaker to be controlled, the environmental noise data of the speaker to be controlled within a preset period is extracted, including the average decibel value of the noise, the equivalent continuous noise level, the peak noise level and the noise spectrum, and the environmental noise evaluation value of the speaker to be controlled is obtained by analysis and processing; The environmental noise evaluation value is used to characterize the degree of influence of the environmental noise on the audio signal to be played; Based on the environmental noise evaluation value of the speaker to be controlled, a mapping and matching is performed with a sound quality enhancement parameter set corresponding to each environmental noise evaluation value interval pre-stored in the database to obtain a sound quality enhancement parameter of the audio signal to be played, and the audio signal is preliminarily enhanced in sound quality based on the sound quality enhancement parameter to obtain an enhanced audio signal; The preset time period size is extracted from the database, and the enhanced audio signal is divided into short time frames, which are recorded as each frame audio signal.

5. The intelligent control method of a loudspeaker audio signal according to claim 1, characterized in that: The step of calculating the frequency domain features of each frame of the audio signal and comprehensively analyzing and processing to generate a clarity evaluation value of the audio signal to be played specifically includes: The frequency domain characteristics of each frame of audio signal are calculated based on fast Fourier transform, including the frequency center, bandwidth and spectral flatness of each frame of audio signal, and the frequency domain characteristics of each frame of audio signal are analyzed and processed to obtain the average bandwidth, frequency center fluctuation rate and average spectral flatness of the audio signal to be played; A set of clarity index parameters is extracted from the database, including bandwidth index, frequency center fluctuation index and spectral flatness index. The average bandwidth, frequency center fluctuation rate and average spectral flatness of the audio signal to be played are compared and analyzed with the clarity index parameter set to obtain a clarity evaluation value of the audio signal to be played.

6. The intelligent control method of a loudspeaker audio signal according to claim 1, characterized in that: The extracting of linear prediction coding coefficients to obtain the formant parameters of each frame of the audio signal and the comprehensive analysis and processing to obtain the audio quality evaluation value of the audio signal to be played specifically include: Calculating the autocorrelation coefficient of each frame of audio signal, and calculating again based on the autocorrelation coefficient of each frame of audio signal to obtain the linear prediction coding coefficient of each frame of audio signal; Based on the linear prediction coding coefficients of each frame of audio signal, the formant parameters of each frame of audio signal are obtained, including the formant position, the formant bandwidth and the formant intensity, and the formant parameters of each frame of audio signal are analyzed and processed to obtain the formant position fluctuation rate, the average formant bandwidth and the formant intensity fluctuation rate of the audio signal to be played; Audio quality index parameters are extracted from a database, including a resonance peak position fluctuation rate index, an average resonance peak bandwidth index and a resonance peak intensity fluctuation rate index. The resonance peak position fluctuation rate, the average resonance peak bandwidth and the resonance peak intensity fluctuation rate of the audio signal to be played are compared and analyzed with the audio quality index parameters to obtain an audio quality evaluation value of the audio signal to be played.

7. The intelligent control method of a loudspeaker audio signal according to claim 1, characterized in that: The comprehensive analysis and processing obtains the audio signal control value set of the audio signal to be played and outputs it after control. The specific processing conditions are: Based on the clarity evaluation value and the audio quality evaluation value of the audio signal to be played, a comprehensive characteristic value of the audio signal to be played is obtained by comprehensive analysis and processing, and the comprehensive characteristic value of the audio signal to be played is mapped and matched with a set of audio signal control parameters corresponding to each comprehensive characteristic value interval pre-stored in the database to obtain a set of control parameters of the audio signal to be played, including a high-frequency control value, a low-frequency control value and an audio compression value; Based on the control parameter set of the audio signal, the audio signal to be played is advanced controlled.

8. The intelligent control method of a loudspeaker audio signal according to claim 7, characterized in that: The comprehensive analysis process obtains the comprehensive characteristic value of the audio signal to be played, and the specific processing conditions are: ; in, is the comprehensive characteristic value of the audio signal to be played, is the clarity evaluation value of the audio signal to be played, is the audio quality evaluation value of the audio signal to be played, is the clarity weight factor, is the audio quality weight factor, is the activation function, .

9. An intelligent control system for loudspeaker audio signals, characterized in that: A trial operation module is used to perform a trial operation on the speaker to be controlled after receiving a start signal, monitor the real-time performance of the voice coil and the real-time performance of the speaker diaphragm of the speaker to be controlled during the trial operation, and obtain the hardware trial operation monitoring result of the speaker to be controlled; The sound quality preliminary enhancement module is used to send a permission formal start instruction when the hardware trial operation monitoring result of the speaker to be controlled is normal, collect the audio signal to be played and the environmental noise of the speaker to be controlled, perform preliminary sound quality enhancement on the audio signal, and divide the enhanced audio signal into short time frames according to a preset time period, which are recorded as each frame audio signal; The audio feature analysis module is used to calculate the frequency domain features of each frame of the audio signal, comprehensively analyze and process to generate the clarity evaluation value of the audio signal to be played, and extract the linear prediction coding coefficient to obtain the formant parameters of each frame of the audio signal, and comprehensively analyze and process to obtain the audio quality evaluation value of the audio signal to be played; The audio control output module is used to comprehensively analyze and process the clarity evaluation value and audio quality evaluation value of the audio signal to be played, obtain the audio signal control value set of the audio signal to be played, and output it after control.

10. A storage medium, characterized in that: include: The storage medium has one or more programs, and the one or more programs are executed by one or more processors to implement the method as claimed in claims 1-8.

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