Control Method, Device and Storage Medium of Audio Processing System

Abstract

The present application discloses a control method, device, and storage medium for an audio processing system. The present application relates to the technical field of audio data processing. The method obtains environmental information and device status data collected by a sensor, then inputs the environmental information and the device status data into the clock compensation module to generate corresponding compensation parameters, and performs a compensation action on the audio clock based on the compensation parameters. Then, the source sampling rate corresponding to at least one audio source is determined according to the compensated audio clock. Finally, the source sampling rate is detected by the sampling rate conversion module. When the source sampling rate is different from the preset sampling rate, the source sampling rate is adjusted based on the preset sampling rate. The present application first compensates the audio clock and uniformly converts the compensated sampling rate through the sampling rate conversion module to achieve clock synchronization between multi-source audio inputs and unify the sampling rates of multi-source audio inputs, so as to improve the processing efficiency of audio data.

Description

Technical Field

[0001] This application relates to the technical field of audio data processing, and particularly to a control method, device, and storage medium for an audio processing system. Background Art

[0002] Traditional audio processing solutions usually adopt fixed sampling rates and clock synchronization mechanisms. When processing audio signals, they mainly rely on preset sampling rates and fixed clock sources. Specifically, it is generally assumed that all audio sources have the same sampling rate, and a central clock is used to synchronize all audio processing operations. In practical applications, different audio sources may have different sampling rates, and clock drift of the device is also inevitable. Due to the lack of flexible clock compensation and sampling rate conversion mechanisms in traditional solutions, the clocks between multi-source audio inputs are out of sync, and the sampling rates of multi-source audio inputs are not unified, thus increasing the system burden, which results in low efficiency when traditional solutions process audio data.

[0003] The above content is only used to assist in understanding the technical solution of this application, and does not represent an admission that the above content is prior art. Summary of the Invention

[0004] This application provides a control method, device, and storage medium for an audio processing system, aiming to solve the problem of low efficiency when traditional solutions process audio data.

[0005] To achieve the above object, a control method for an audio processing system provided by this application is applied to an audio processing system. The audio processing system includes a clock compensation module and a sampling rate conversion module. The control method of the audio processing system includes the following steps:

[0006] Obtain environmental information and device status data collected by a sensor;

[0007] Input the environmental information and the device status data into the clock compensation module to generate corresponding compensation parameters, and perform a compensation action on the audio clock based on the compensation parameters;

[0008] Determine the source sampling rate corresponding to at least one audio source according to the compensated audio clock;

[0009] Detect the source sampling rate through the sampling rate conversion module. When the source sampling rate is different from the preset sampling rate, adjust the source sampling rate based on the preset sampling rate.

[0010] In one embodiment, before the step of obtaining environmental information and device status data collected by a sensor, the method further includes:

[0011] Obtain an initial timestamp recorded based on the system clock through the clock compensation module, and determine the time difference according to the current clock value;

[0012] Determine the audio source sampling rate corresponding to the initial timestamp, and determine the number of audio samples corresponding to the audio source sampling rate according to the time difference;

[0013] Obtain the actual number of samples corresponding to the buffer, compare the actual number of samples with the number of audio samples, and if they are different, perform corresponding compensation actions.

[0014] In one embodiment, the step of performing corresponding compensation actions includes:

[0015] Determine the ratio of the audio source sampling rate to the actual sampling rate according to the audio source sampling rate and the actual sampling rate corresponding to the actual number of samples;

[0016] Determine the target reading speed according to the ratio and the number of samples corresponding to the reading speed;

[0017] Adjust the target reading speed to the current reading speed.

[0018] In one embodiment, the step of detecting the source sampling rate by the sampling rate conversion module and adjusting the source sampling rate based on the preset sampling rate when the source sampling rate is different from the preset sampling rate includes:

[0019] Obtain the preset sampling rate;

[0020] Compare the source sampling rate and the preset sampling rate through the sampling rate conversion module;

[0021] When the source sampling rate is less than the preset sampling rate, determine the proportional relationship according to the source sampling rate and the preset sampling rate;

[0022] Determine the sample points to be inserted into the source sampling rate according to the proportional relationship, and insert the sample points into the source sampling rate based on the preset interpolation type.

[0023] In one embodiment, after the step of determining the sample points to be inserted into the source sampling rate according to the proportional relationship and inserting the sample points into the source sampling rate based on the preset interpolation type, it further includes:

[0024] When the source sampling rate is greater than the preset sampling rate, determine the proportional relationship between the source sampling rate and the preset sampling rate;

[0025] Downsample the source sampling rate according to the proportional relationship

[0026] In one embodiment, after the step of detecting the source sampling rate by the sampling rate conversion module and adjusting the source sampling rate based on the preset sampling rate when the source sampling rate is different from the preset sampling rate, the following steps are further included:

[0027] Detect system resource information through the audio delay control module;

[0028] When the system resource information is greater than or equal to a preset threshold, set the read / write speed of the audio data to a first preset speed to reduce the read / write beats of the audio data;

[0029] When the system resource information is less than the preset threshold, set the read / write speed of the audio data to a second preset speed.

[0030] In one embodiment, after the step of detecting system resource information through the audio delay control module, the following steps are further included:

[0031] Obtain the audio tasks to be processed in the queue to be processed and determine the priorities of the audio tasks to be processed;

[0032] When the system resource information is greater than or equal to a preset threshold, determine target audio tasks with priorities higher than the preset priority among the audio tasks to be processed;

[0033] Place the target audio tasks in the priority processing position in the queue to be processed.

[0034] In one embodiment, before the step of detecting system resource information through the audio delay control module, the following steps are further included:

[0035] Collect the audio delay, buffer size, and remaining data volume in the audio processing process based on the audio delay control module;

[0036] Input the audio processing information and resource usage information into a predictive read / write scheduling model and determine the prediction result output by the predictive read / write scheduling model;

[0037] Generate an audio processing strategy according to the prediction result and optimize the audio processing process based on the audio processing strategy.

[0038] In addition, to achieve the above object, the present application further provides a control device for an audio processing system. The control device for the audio processing system includes: a memory, a processor, and a computer program stored on the memory and executable on the processor. The computer program is configured to implement the steps of the control method for the audio processing system as described above.

[0039] In addition, to achieve the above object, the present application further provides a storage medium, which is a computer-readable storage medium. A computer program is stored on the computer-readable storage medium. When the computer program is executed by a processor, the steps of the control method of the audio processing system as described above are implemented.

[0040] The present application provides a control method for an audio processing system, a control device for an audio processing system, and a storage medium. By acquiring environmental information and device status data collected by a sensor, then inputting the environmental information and the device status data into the clock compensation module to generate corresponding compensation parameters, and performing a compensation action on the audio clock based on the compensation parameters, and then determining the source sampling rate corresponding to at least one audio source according to the compensated audio clock, and finally detecting the source sampling rate through the sampling rate conversion module. When the source sampling rate is different from the preset sampling rate, the source sampling rate is adjusted based on the preset sampling rate. The present application compensates the audio clock through the clock compensation module and uniformly converts the compensated sampling rate through the sampling rate conversion module to improve the processing efficiency of audio data. BRIEF DESCRIPTION OF THE DRAWINGS

[0041] The accompanying drawings herein are incorporated into the specification and form a part of the specification, showing embodiments consistent with the present application, and are used together with the specification to explain the principles of the present application.

[0042] To more clearly illustrate the technical solutions in the embodiments of the present application or the prior art, the following will briefly introduce the accompanying drawings required for use in the description of the embodiments or the prior art. Obviously, for those of ordinary skill in the art, other drawings can also be obtained based on these drawings without creative efforts.

[0043] Figure 1 It is a schematic flowchart of the first embodiment of the control method for the audio processing system of the present application;

[0044] Figure 2 It is a schematic flowchart of the second embodiment of the control method for the audio processing system of the present application;

[0045] Figure 3 It is a schematic flowchart of the third embodiment of the control method for the audio processing system of the present application;

[0046] Figure 4 It is a schematic architecture diagram of the hardware operating environment of the control device for the audio processing system involved in the embodiments of the present application.

[0047] The implementation, functional features, and advantages of the object of the present application will be further described with reference to the embodiments and the accompanying drawings. DETAILED DESCRIPTION OF THE EMBODIMENTS

[0048] It should be understood that the specific embodiments described herein are merely for explaining the present application and are not intended to limit the present application.

[0049] To better understand the above technical solution, the exemplary embodiments of the present application will be described in more detail below with reference to the accompanying drawings. Although the exemplary embodiments of the present application are shown in the drawings, it should be understood that the present application can be implemented in various forms and should not be limited by the embodiments set forth herein. On the contrary, these embodiments are provided to enable a more thorough understanding of the present application and to fully convey the scope of the present application to those skilled in the art.

[0050] To better understand the above technical solution, the above technical solution will be described in detail below in conjunction with the accompanying drawings of the specification and specific implementation manners.

[0051] The main solution of the present application is: obtaining environmental information and device status data collected by a sensor; inputting the environmental information and the device status data into the clock compensation module to generate corresponding compensation parameters, and performing a compensation action on the audio clock based on the compensation parameters; determining the source sampling rate corresponding to at least one audio source according to the compensated audio clock; detecting the source sampling rate through the sampling rate conversion module, and when the source sampling rate is different from a preset sampling rate, adjusting the source sampling rate based on the preset sampling rate.

[0052] Traditional audio processing solutions usually adopt fixed sampling rates and clock synchronization mechanisms. When processing audio signals, they mainly rely on preset sampling rates and fixed clock sources. Specifically, it is generally assumed that the sampling rates of all audio sources are the same, and a central clock is used to synchronize all audio processing operations. In practical applications, different audio sources may have different sampling rates, and clock drift of the device is also inevitable. Due to the lack of flexible clock compensation and sampling rate conversion mechanisms in traditional solutions, the clocks between multi-source audio inputs are not synchronized, and the sampling rates of multi-source audio inputs are not unified, thus increasing the system burden, which results in low efficiency when traditional solutions process audio data.

[0053] By obtaining environmental information and device status data collected by a sensor, then inputting the environmental information and the device status data into the clock compensation module to generate corresponding compensation parameters, and performing a compensation action on the audio clock based on the compensation parameters, then determining the source sampling rate corresponding to at least one audio source according to the compensated audio clock, and finally detecting the source sampling rate through the sampling rate conversion module, and when the source sampling rate is different from a preset sampling rate, adjusting the source sampling rate based on the preset sampling rate. The present application compensates the audio clock through the clock compensation module and uniformly converts the compensated sampling rate through the sampling rate conversion module to improve the processing efficiency of audio data.

[0054] It should be noted that the execution entity of this embodiment can be an audio processing system, or a computing service device with data processing, network communication, and program running functions, such as a tablet computer, a personal computer, a mobile phone, etc., or a control device of an audio processing system that can implement the above functions. This embodiment does not make specific limitations in this regard. Hereinafter, taking the audio processing system as the execution entity as an example, this embodiment and the following embodiments will be described.

[0055] Embodiment 1

[0056] Based on this, an embodiment of the present application provides a control method for an audio processing system. Refer to Figure 1 , Figure 1 which is a schematic flowchart of the first embodiment of the control method for the audio processing system of the present application. The control method for the audio processing system includes steps S10 to S40:

[0057] Step S10: Obtain the environmental information and device status data collected by the sensor.

[0058] In this embodiment, the audio processing system performs processing actions. The audio processing system includes a clock compensation module and a sampling rate conversion module. The sensor includes a temperature sensor, a humidity sensor, and a pressure sensor. The sensor is used to detect and collect the environmental information and device status data of the audio processor. Among them, the audio processing system is deployed in the audio processor. The environmental information includes environmental temperature, humidity, and air pressure. The device status data includes device load, such as CPU usage rate, memory occupancy rate, etc., and also includes device running time.

[0059] After the sensor collects the environmental information and device status data, the data is transmitted to the audio processing system through the Internet of Things technology, that is, through the sensor network.

[0060] Specifically, the sensor converts the collected physical quantity into a measurable electrical signal, and performs preliminary processing and data formatting through an internal processing unit. Then the processed data is prepared for transmission through the communication unit and sent to the network layer. According to the application scenario and requirements, a suitable wireless communication technology is selected, such as WiFi, Bluetooth, Zigbee, LoRaWAN, NB-IoT, etc. Then the sensor node sends the data to the converged network of the communication network and the Internet in the network layer through the selected wireless communication technology. Receiving devices in the network layer, such as routers, base stations, etc., receive the data from the sensor node and forward it to the audio processing system. The audio processing system receives the data from the network layer through the network interface, thereby obtaining the environmental information and device status data collected by the sensor.

[0061] Step S20: Input the environmental information and the device status data into the clock compensation module to generate corresponding compensation parameters, and perform a compensation action on the audio clock based on the compensation parameters.

[0062] In this embodiment, the clock compensation module is a module used to correct the problem of audio clock asynchronization in an audio processing system. It can adjust the audio clock according to the input relevant data to reduce the impact caused by clock asynchronization. The audio clock refers to the clock signal used to control operations such as audio signal sampling and playback in an audio processing system, and its accuracy directly affects the audio quality. By inputting the environmental information and the device status data into the clock compensation module, the compensation parameters are calculated and generated based on the clock compensation module, and finally a compensation action is performed on the audio clock based on the compensation parameters, that is, the audio clock is adjusted to ensure the synchronization of the audio signal.

[0063] Specifically, the audio processing system inputs the environmental information collected by the sensor and the device status data into the clock compensation module. The clock compensation module calculates the clock offset value based on these data. The system adjusts the audio clock according to the clock offset value. For example, assume that the system detects an offset in the audio clock and the calculated offset value is 0.1 second. Then the clock compensation module adjusts the audio clock according to the offset value.

[0064] As an optional implementation manner of calculating the compensation parameters, the compensation parameters can be output by a preset training model based on the environmental information and the device status data. The preset training model is based on a neural network to learn the influence of external environmental changes on the audio clock. The inputs of the neural network can be sensor data such as environmental temperature and humidity (these factors may affect the performance of the electronic components of the audio device, and thus affect the clock accuracy), and information such as device load (such as CPU usage rate, memory occupancy rate, etc.). In the training stage, by artificially creating different environmental conditions and device load situations, and simultaneously recording the corresponding audio clock deviation situations, these data are used as training samples. For example, when the environmental temperature rises, the crystal oscillator frequency of the audio device may change slightly, resulting in a clock deviation. The neural network learns these samples to establish a relationship model between the environment and the device status and the clock deviation, that is, the preset training model.

[0065] In actual operation, when new environmental and device status data are received, the neural network can quickly predict the clock deviation according to the learned model and generate corresponding compensation parameters. In addition, a reward mechanism can be defined through a reinforcement learning algorithm to optimize the clock compensation strategy. The agent (representing the compensation strategy) can take different actions, such as adjusting the clock frequency, changing the buffer size, etc. By continuously trying different action combinations and learning according to the reward feedback, the reinforcement learning algorithm can find the optimal compensation strategy.

[0066] As an alternative implementation for generating compensation parameters, it can be calculated by means of spline interpolation. First, collect clock offset data at known time points in the database. These data points should be as evenly distributed as possible to ensure the accuracy of the interpolation result. Then, according to the characteristics and requirements of the data, select a spline function. Common spline functions include linear spline, quadratic spline, and cubic spline, etc. Among them, cubic spline interpolation has higher smoothness and accuracy, and cubic spline interpolation is usually used as the spline function. Then, use the known data points and the spline function to construct a spline curve. Finally, at the time point where prediction is needed, use the constructed spline curve to estimate the clock offset. Thus, the calculation of the clock offset amount is achieved by calculating the function value of the spline curve at this time point.

[0067] Step S30: Determine the source sampling rate corresponding to at least one audio source according to the compensated audio clock.

[0068] In this embodiment, the source sampling rate refers to the original sampling rate of the audio source, that is, the sampling frequency when the audio signal is collected. The compensated audio clock can be used to determine the source sampling rate of the audio source, and the source sampling rate of the audio source can be determined through the frequency of the audio clock. There is at least one audio source, and by using an ARM chip as the audio processing chip, 16 channels of analog, 16 channels of Dante / network, and 4 channels of USB, a total of 36 channels of audio input can be supported.

[0069] Specifically, the audio processing system analyzes the spectrum of the audio signal according to the compensated audio clock to determine the source sampling rate. Technologies such as Fourier transform can be used to analyze the frequency components of the audio signal, thereby determining the source sampling rate. The audio signal is usually a continuous analog signal. In order to process it on a computer, it needs to be sampled and converted into a discrete signal. The selection of the sampling frequency is crucial and must satisfy the Nyquist sampling theorem, that is, the sampling frequency should be greater than twice the highest frequency of the signal to avoid aliasing. Preprocess the sampled signal, such as removing the DC component, normalizing, etc., to improve the accuracy of subsequent analysis. Use Fourier transform technology to convert the audio signal in the time domain into a frequency domain representation. This step is the core of spectrum analysis, which allows us to observe the energy distribution of the signal at different frequencies. Analyze the frequency domain signal after Fourier transform to identify each frequency component in the signal. These components are usually represented in complex form, containing amplitude and phase information. By analyzing features such as peaks and valleys in the spectrogram, the frequency components, harmonic structure, etc. of the signal can be understood. Based on the results of spectrum analysis, combined with known audio signal processing algorithms and features, the original sampling rate can be inferred.

[0070] Step S40: Detect the source sampling rate through the sampling rate conversion module. When the source sampling rate is different from the preset sampling rate, adjust the source sampling rate based on the preset sampling rate.

[0071] In this embodiment, the sampling rate conversion module is used to convert an audio signal from one sampling rate to another. Through interpolation and filtering techniques, the quality of the audio signal can be ensured during the conversion process. Specifically, the sampling rate can be converted by means of sinc interpolation, anti-aliasing filters, etc. The audio processing system detects the source sampling rate of the audio source through the sampling rate conversion module. If the source sampling rate is different from the preset sampling rate, the audio processing system uses the sampling rate conversion module to make adjustments.

[0072] Specifically, according to the set preset sampling rate, digital signal processing algorithms such as interpolation and filtering are used. When the source sampling rate is lower than the preset sampling rate, sample points are supplemented through interpolation to improve the audio resolution; when it is higher than the preset sampling rate, the sampling rate is reasonably downsampled and redundant high-frequency components are filtered out to avoid aliasing, so that all audio streams flow at a uniform sampling rate.

[0073] Optionally, in this embodiment, the step of detecting the source sampling rate through the sampling rate conversion module and adjusting the source sampling rate based on the preset sampling rate when the source sampling rate is different from the preset sampling rate includes:

[0074] Obtain the preset sampling rate; compare the source sampling rate and the preset sampling rate through the sampling rate conversion module; when the source sampling rate is less than the preset sampling rate, determine the proportional relationship according to the source sampling rate and the preset sampling rate; determine the sample points that need to be inserted into the source sampling rate according to the proportional relationship, and insert the sample points into the source sampling rate based on the preset interpolation type.

[0075] Specifically, the preset sampling rate is the sampling frequency preset in the audio processing system. It is determined based on factors such as the playback device of the system and audio processing requirements, and is used to ensure the unified processing and high-quality output of audio signals in the system. During the configuration phase of the audio processing system, a suitable sampling rate is determined as the preset sampling rate according to the target application scenario and device performance, and is stored in the parameter settings of the system. The sampling rate conversion module first reads the source sampling rate of the audio source, and then compares it with the preset sampling rate stored in the audio processing system. This comparison process usually involves simple numerical comparison to determine the relationship between the two sampling rates, such as whether they are equal, which one is higher or lower. When the sampling rate conversion module determines that the source sampling rate is less than the preset sampling rate, the ratio of the two sampling rates is calculated. For example, if the source sampling rate is 32 kHz and the preset sampling rate is 44.1 kHz, the ratio is 44.1 / 32. This ratio will be used in subsequent sample point insertion operations to ensure that the number and position of the inserted sample points can smoothly transition the source sampling rate to the preset sampling rate.

[0076] Interpolation is a mathematical method used to estimate the values of unknown data points between known data points. The preset interpolation type refers to the interpolation algorithm preselected during the sampling rate conversion process, such as linear interpolation, spline interpolation, etc., and is used to generate new sample points. According to the ratio calculated in the above steps, the number of sample points to be inserted between each sampling point of the source sampling rate is determined. Then, the preset interpolation type (such as linear interpolation) is used to calculate the values of these new sample points. For example, if the ratio is 1.378125, it may be necessary to insert 0.378125 sample points between every two source sample points, and the specific values of these new sample points are calculated through the interpolation algorithm, thus completing the insertion of the sample points into the source sampling rate based on the preset interpolation type.

[0077] Furthermore, in this embodiment, after the step of determining the sample points to be inserted into the source sampling rate according to the ratio and inserting the sample points into the source sampling rate based on the preset interpolation type, the following steps are further included:

[0078] When the source sampling rate is greater than the preset sampling rate, determine the ratio relationship between the source sampling rate and the preset sampling rate; downsample the source sampling rate according to the ratio relationship.

[0079] Specifically, when the sampling rate conversion module detects that the source sampling rate is higher than the preset sampling rate, it calculates the ratio of the two sampling rates. This ratio will be used in the subsequent downsampling operation to determine how many sample points need to be discarded or how to adjust the interval between sample points. Downsampling refers to the process of reducing the sampling rate of an audio signal, usually by discarding certain sample points or using a filter to reduce the number of sample points while trying to maintain the quality of the audio signal. According to the determined proportional relationship, the sampling rate conversion module will perform downsampling on the source sampling rate. If the proportional relationship is an integer, some sample points can be directly discarded at regular intervals; if the proportional relationship is a decimal, filtering and interpolation techniques are used to smoothly reduce the sample points.

[0080] In addition, the quality of the audio signal needs to be considered. A low-pass filter is usually used to prevent aliasing effects and ensure that the downsampled audio signal still maintains good quality. For example, during the downsampling process from 96 kHz to 48 kHz, the sampling rate conversion module will first apply a low-pass filter with a cut-off frequency set at 24 kHz (half of 48 kHz) to filter out frequency components above 24 kHz, and then perform the operation of discarding sample points. This can effectively reduce the audio distortion that may occur during the downsampling process.

[0081] Optionally, for speech audio sources, since the frequency range of speech signals is relatively narrow, when performing sampling rate conversion, more attention can be paid to retaining the clarity and intelligibility of speech. When mapping a speech audio source from a lower sampling rate to the target sampling rate, a linear interpolation method can be used to fill in the new sample points to avoid generating excessive harmonic distortion. If the target sampling rate is an integer multiple of the source sampling rate, a new sample point can be simply inserted between every two original sample points, and its value can be calculated through linear interpolation. At the same time, since speech signals are relatively sensitive to phase changes, attention should be paid to maintaining the phase continuity of speech signals during the sampling rate conversion process to ensure the naturalness of speech.

[0082] Optionally, music audio sources have rich harmonic components. When performing sampling rate conversion, more attention needs to be paid to retaining the timbre and rhythm of music. Using a high-quality sinc-function-based interpolation algorithm, new sample points can be calculated by weighted summation of the original audio signal, where the weights are determined by the sinc-function. This algorithm can better handle the high-frequency components in music signals and reduce the degradation of audio quality caused by sampling rate conversion, especially for music played by musical instruments with complex timbres.

[0083] In the technical solution provided in this embodiment, by acquiring the environmental information and device status data collected by the sensor, then inputting the environmental information and the device status data into the clock compensation module to generate corresponding compensation parameters, performing a compensation action on the audio clock based on the compensation parameters, determining the source sampling rate corresponding to at least one audio source according to the compensated audio clock, and finally detecting the source sampling rate through the sampling rate conversion module. When the source sampling rate is different from the preset sampling rate, the source sampling rate is adjusted based on the preset sampling rate. The solution of this embodiment compensates the audio clock through the clock compensation module and uniformly converts the sampling rate after compensation through the sampling rate conversion module, so as to synchronize the clocks between multi-source audio inputs and unify the sampling rates of multi-source audio inputs, thereby improving the processing efficiency of audio data.

[0084] Embodiment Two

[0085] Please refer to Figure 2 , in the second embodiment, before the steps of step S10, there are also steps S50 to S70:

[0086] Step S50: Obtain the initial timestamp recorded based on the system clock through the clock compensation module, and determine the time difference according to the current clock value.

[0087] In this embodiment, the initial timestamp refers to a time reference point recorded according to the system clock at the start of audio processing, which is used for subsequent time calculation and synchronization. The time difference refers to the time interval between the current clock value and the initial timestamp. At the start of audio processing, the clock compensation module obtains the current value of the system clock and records it as the initial timestamp. During the audio processing, the clock compensation module continuously detects the change of the system clock, obtains the current clock value, and determines the time difference according to the current clock value and the initial timestamp. This time difference will be used for subsequent audio sample number calculation and execution of compensation actions.

[0088] Step S60: Determine the audio source sampling rate corresponding to the initial timestamp, and determine the number of audio samples corresponding to the audio source sampling rate according to the time difference.

[0089] In this embodiment, the audio source sampling rate refers to the sampling frequency used by the audio source (such as an audio file, microphone input, etc.) when collecting or storing an audio signal, which determines the number of sampling points of the audio signal per unit time. The number of audio samples refers to the number of sampling points of the audio signal calculated according to the audio source sampling rate within a given time interval. This parameter is very important for audio playback, processing, and synchronization.

[0090] Specifically, according to the initial timestamp, determine the sampling rate of the audio source at that time. This is usually a parameter known at the beginning of audio processing or can be obtained from the metadata of the audio file. Using the calculated time difference and combining it with the audio source sampling rate, calculate the number of audio samples that should theoretically be generated within this time difference through the formula "number of audio samples = time difference × audio source sampling rate".

[0091] Step S70: Obtain the actual number of samples corresponding to the buffer, compare the actual number of samples with the number of audio samples. If they are different, perform corresponding compensation actions.

[0092] In this embodiment, the buffer refers to the area in the audio processing system used to temporarily store audio data, which plays a role in buffering and synchronization during the reading, processing, and playing of audio. The actual number of samples refers to the number of audio samples actually stored in the buffer, and this number may be inconsistent with the theoretically calculated number of audio samples due to factors such as clock deviation and processing delay. The compensation action refers to a series of adjustment measures taken to correct problems such as clock deviation or inconsistent number of samples during audio processing, such as inserting or deleting sample points, adjusting the playback speed, etc., to ensure the accurate playback and synchronization of audio.

[0093] Obtain the currently actual stored number of audio samples from the buffer of the audio processing system. Compare the actual number of samples with the calculated theoretical number of audio samples. If the two are different, perform corresponding compensation actions according to the difference. If the actual number of samples is less than the theoretical number of samples, sample points need to be inserted to supplement; if the actual number of samples is more than the theoretical number of samples, sample points need to be deleted or the playback speed needs to be adjusted to make the audio playback consistent with the theoretical time sequence.

[0094] Optionally, use the system high-precision clock (CLOCK_MONOTONIC) to record the initial timestamp as the "time zero point" of the audio source. Subsequently, regularly obtain the current clock value again and calculate the elapsed time through the time difference. According to the known sampling rate of the audio source, convert the elapsed time into the number of audio samples that should theoretically be collected. Compare the number of samples in the actual buffer. If there is a deviation (caused by factors such as device clock accuracy differences and transmission delay fluctuations), dynamically adjust the indexes, writing positions, etc. of subsequent data processing links to ensure that each audio stream is aligned in the time dimension and compensate for the time misalignment caused by hardware differences.

[0095] Optionally, in this embodiment, the step of performing corresponding compensation actions includes:

[0096] Determine the ratio of the audio source sampling rate to the actual sampling rate corresponding to the actual number of samples; determine the target reading speed according to the ratio and the number of samples corresponding to the reading speed; adjust the target reading speed to the current reading speed.

[0097] Specifically, determine the ratio of the audio source sampling rate to the actual sampling rate corresponding to the actual number of samples, then determine the target reading speed according to the ratio and the number of samples corresponding to the reading speed, and adjust the target reading speed to the current reading speed.

[0098] Exemplarily, the audio source feature sampling rate is detected in real time through an intelligent algorithm. When the system receives an audio source with a sampling rate of 48 kHz, the algorithm will record this parameter. If the sampling rate of the audio source later becomes 44.1 kHz, the algorithm can calculate the compensation parameter according to the pre-set rules. Adjust the reading speed of the audio buffer by calculating the ratio of the two (44.1 / 48). If the original reading speed was to read 100 audio samples per second, it is now adjusted to read 100*(44.1 / 48)≈92 audio samples per second to achieve audio clock synchronization.

[0099] In the technical solution provided in this embodiment, the clock compensation module obtains the initial timestamp recorded based on the system clock, determines the time difference according to the current clock value, then determines the audio source sampling rate corresponding to the initial timestamp, determines the number of audio samples corresponding to the audio source sampling rate according to the time difference, and then obtains the actual number of samples corresponding to the buffer. Compare the actual number of samples with the number of audio samples. If they are different, perform the corresponding compensation action. The solution of this embodiment determines the time difference through the initial timestamp recorded by the clock compensation module based on the system clock, and then compensates the audio clock according to the determined number of audio samples to achieve audio clock synchronization.

[0100] Embodiment Three

[0101] Please refer to Figure 3 , in the third embodiment, after the step of detecting the source sampling rate by the sampling rate conversion module and adjusting the source sampling rate based on the preset sampling rate when the source sampling rate is different from the preset sampling rate, steps S80 to S100 are further included:

[0102] Step S80: Detect the system resource information through the audio delay control module.

[0103] In this embodiment, the audio delay control module is used to manage and control the audio signal delay, capable of detecting the usage of system resources, and adjusting audio processing parameters according to the resource status to optimize audio performance and system stability. System resource information refers to the available resource data in a computer or audio processing system, such as CPU usage rate, memory occupancy, disk I / O speed, etc. These information reflect the current running state and processing capacity of the system.

[0104] Specifically, the audio delay control module starts and initializes to prepare for detecting system resources. It obtains system resource information such as CPU usage rate, memory occupancy, disk I / O speed, etc. from the system resource manager or relevant interfaces in real time. The obtained system resource information is stored in an internal data structure.

[0105] It is also possible to build an ALSA (Advanced Linux Sound Architecture) read / write scheduling mechanism linked to system load perception, which detects the CPU usage rate, remaining memory, and the busy degree of other processes in real time. When the system resources are abundant, it increases the ALSA read / write frequency, speeds up the audio data in and out speed, reduces the duration of data backlog waiting in the buffer, and reduces the delay; when the system is busy, it intelligently reduces the read / write rhythm to avoid over-preempting resources and causing system jamming and audio disconnection, and optimizes the delay characteristics on the premise of ensuring overall stability. Assign priorities to different application scenarios.

[0106] Step S90: When the system resource information is greater than or equal to a preset threshold, set the read / write speed of the audio data to a first preset speed to reduce the read / write rhythm of the audio data.

[0107] In this embodiment, before increasing the read / write speed, the audio processing system will pre-read a certain amount of audio data into the buffer to ensure the smoothness of audio playback. The preset threshold refers to a critical value of the system resource usage rate set in advance. When the system resource information exceeds this value, it indicates that the system may be in a high-load state, and measures need to be taken to reduce the processing speed of the audio data to avoid system overload. The first preset speed refers to a slower read / write speed set to reduce the read / write rhythm of the audio data when the system resources are tense, aiming to reduce the occupancy of system resources by audio processing and ensure the stable operation of the system. The read / write rhythm of the audio data refers to the time interval or frequency during the reading and writing of the audio data. The faster the read / write rhythm, the larger the amount of audio data processed per unit time, and the higher the requirement for system resources.

[0108] Specifically, the audio delay control module compares the obtained system resource information with a preset threshold. If the system resource information is greater than or equal to the preset threshold, indicating that the system resources are tight, the module adjusts the read / write speed of the audio data to the first preset speed. By adjusting the read / write speed, the read / write beats of the audio data are reduced, and the occupation of system resources by audio processing is decreased, thereby alleviating the load pressure on the system.

[0109] Step S100: When the system resource information is less than the preset threshold, set the read / write speed of the audio data to the second preset speed.

[0110] In this embodiment, the second preset speed refers to a relatively fast read / write speed set to improve the audio processing efficiency and performance when the system resources are sufficient, with the aim of fully utilizing the system resources on the premise of ensuring the stable operation of the system.

[0111] Specifically, the audio delay control module continuously detects the system resource information and compares it with the preset threshold. If the system resource information is less than the preset threshold, indicating that the system resources are relatively sufficient, the module adjusts the read / write speed of the audio data to the second preset speed. By increasing the read / write speed, the processing speed of the audio data is accelerated, the smoothness and response speed of audio playback are improved, and at the same time, the system resources are fully utilized to improve the overall performance.

[0112] Optionally, in this embodiment, after the step of detecting the system resource information by the audio delay control module, it further includes:

[0113] Obtain the to-be-processed audio tasks in the to-be-processed queue and determine the priorities of the to-be-processed audio tasks; when the system resource information is greater than or equal to the preset threshold, determine the target audio tasks with priorities higher than the preset priority among the to-be-processed audio tasks; place the target audio tasks in the priority processing position in the to-be-processed queue.

[0114] Specifically, the priorities are dynamically adjusted according to the real-time status and importance of the application programs. For example, during a live performance, if it is found that a certain audio effect processing task is crucial for the performance effect and there is a risk of audio interruption for this task, then the priority of this task can be temporarily increased to ensure that it can obtain resources in time and complete the processing. For high-real-time scenarios, such as audio processing during a live performance, the highest priority is assigned. When the system resources are tight, the scheduler of the operating system will first ensure the execution of high-priority tasks.

[0115] For example, in a system that runs multiple audio applications simultaneously (including live performance audio processing, music playback, and voice recording), the priority of live performance audio processing is set to the highest. When CPU and memory resources are strained, the scheduler will pause low-priority tasks such as music playback and voice recording and preferentially allocate resources to live performance audio processing to ensure that the audio of live performances can be processed and played in real time with high quality.

[0116] In addition, leveraging a multi-core processor architecture, independent audio processing subtasks (such as audio filtering for different channels and multi-band sound effect adjustment) are allocated to different cores for parallel execution; meanwhile, a pipelined processing flow is designed to enable the data processed in the previous stage to seamlessly flow into the next stage. For example, immediately after audio acquisition, it enters the low-latency ASRC (Asynchronous Sample Rate Conversion) stage, and as soon as the conversion is completed, it enters the mixing process, eliminating idle waiting time throughout the process, minimizing the overall processing latency to the greatest extent, and improving the real-time response ability of the audio.

[0117] Insert delay detection probes at each node of ALSA initialization, read / write, and audio processing to capture in real time the time consumed by audio data staying and being processed at each stage, and aggregate them into a complete delay analysis map. This accurately locates the delay bottleneck, whether it is due to improper device initialization configuration, poor read / write scheduling, or sluggish algorithm processing, providing data support for targeted optimization.

[0118] Optionally, in this embodiment, before the step of detecting system resource information through the audio delay control module, the following steps are further included:

[0119] Based on the audio delay control module, collect the audio delay, buffer size, and remaining data volume in the audio processing process; input the audio processing information and resource usage information into a predictive read / write scheduling model, and determine the prediction result output by the predictive read / write scheduling model; generate an audio processing strategy according to the prediction result, and optimize the audio processing process based on the audio processing strategy.

[0120] Specifically, construct a predictive scheduling model based on the historical transmission law of audio data and the characteristics of application scenarios (such as the periodicity of music rhythm and the intermittency of voice calls). Anticipate the audio data volume and transmission requirements in the next stage in advance, and schedule ALSA pre-reading / writing operations in advance during low-load periods to make the audio data ready in advance, filling potential processing gaps to ensure that there is sufficient data to respond in a timely manner at moments with high real-time requirements, such as when playing complex rhythm passages in music or when multiple parties are speaking simultaneously during a call, effectively controlling the sudden increase in delay.

[0121] Based on the feedback data of the detection tool, an adaptive adjustment mechanism is constructed. Once it is found that the delay of a certain link exceeds the standard, the corresponding optimization strategy is automatically triggered, such as adjusting the ALSA buffer parameters, switching to the lightweight algorithm mode, or rebalancing the read-write scheduling priority. Continuously iterate and optimize in a closed loop to ensure that the audio delay is always maintained at a controllable and ideal low-delay level in the complex and changing operating environment of the audio processing system and diverse audio application scenarios, realizing the efficient and stable operation of the system and excellent audio experience.

[0122] Exemplarily, during an online meeting, various relevant data are collected, such as audio delay, buffer size and remaining data volume, device CPU and memory usage, etc. as the input of the model. Machine learning algorithms, such as long short-term memory network (LSTM), are used to build a delay prediction model to learn the patterns in historical data and predict the future change trend of the delay. In the training stage, the audio delay, buffer size and remaining data volume in the past few minutes are used as training samples to let LSTM learn the relationships between these data. After the model training is completed, the current audio delay, buffer size and remaining data volume are input into the model, and the model can predict the delay situation in the next period of time (such as the next 5 seconds), and take measures in advance to ensure the audio output.

[0123] In the technical solution provided in this embodiment, the system resource information is detected by the audio delay control module. When the system resource information is greater than or equal to the preset threshold, the read-write speed of the audio data is set to the first preset speed to reduce the read-write rhythm of the audio data. When the system resource information is less than the preset threshold, the read-write speed of the audio data is set to the second preset speed. Thereby ensuring the high real-time performance of audio signal processing.

[0124] Since the system introduced in the embodiments of this application is the system adopted for implementing the method in the embodiments of this application, based on the method introduced in the embodiments of this application, those skilled in the art can understand the specific structure and variations of the system, so it will not be elaborated here. Any system adopted by the method in the embodiments of this application belongs to the scope to be protected by this application.

[0125] This application provides a control device for an audio processing system. The control device of the audio processing system includes: at least one processor; and a memory communicatively connected to the at least one processor; wherein, the memory stores instructions executable by the at least one processor, and the instructions are executed by the at least one processor to enable the at least one processor to execute the control method of the audio processing system in the first embodiment above.

[0126] Next, refer to Figure 4, which shows a schematic structural diagram of a control device suitable for implementing the audio processing system according to the embodiments of the present application. The control device of the audio processing system in the embodiments of the present application may include, but is not limited to, mobile terminals such as mobile phones, laptop computers, digital broadcast receivers, personal digital assistants (PDAs), tablet computers (PADs), portable multimedia players (PMPs), vehicle terminals (such as vehicle navigation terminals), etc., and fixed terminals such as digital TVs, desktop computers, etc. Figure 4 The control device of the audio processing system shown is only an example and should not impose any limitation on the functions and usage scope of the embodiments of the present application.

[0127] As Figure 4 shown, the control device of the audio processing system may include a processing device 1001 (such as a central processing unit, a graphics processing unit, etc.), which may perform various appropriate actions and processes according to a program stored in a read-only memory (ROM) 1002 or a program loaded from a storage device 1003 into a random access memory (RAM) 1004. In the RAM 1004, various programs and data required for the operation of the control device of the audio processing system are also stored. The processing device 1001, the read-only memory 1002, and the random access memory 1004 are connected to each other through a bus 1005. An input / output (I / O) interface 1006 is also connected to the bus. Generally, the following systems may be connected to the I / O interface 1006: an input device 1007 including, for example, a touch screen, a touch pad, a keyboard, a mouse, an image sensor, a microphone, an accelerometer, a gyroscope, etc.; an output device 1008 including, for example, a liquid crystal display (LCD), a speaker, a vibrator, etc.; a storage device 1003 including, for example, a magnetic tape, a hard disk, etc.; and a communication device 1009. The communication device 1009 may allow the control device of the audio processing system to communicate with other devices wirelessly or wiredly to exchange data. Although the figure shows a control device of an audio processing system having various systems, it should be understood that it is not required to implement or have all the systems shown. More or fewer systems may be implemented or had alternatively.

[0128] In particular, according to the embodiments disclosed in the present application, the processes described above with reference to the flowcharts can be implemented as computer software programs. For example, the embodiments disclosed in the present application include a computer program product that includes a computer program carried on a computer-readable medium, and the computer program contains program codes for executing the methods shown in the flowcharts. In such an embodiment, the computer program can be downloaded and installed from a network through a communication device, or installed from a storage device 1003, or installed from a ROM 1002. When the computer program is executed by a processing device 1001, the above-mentioned functions defined in the methods of the embodiments disclosed in the present application are executed.

[0129] The control device of the audio processing system provided by the present application adopts the control method of the audio processing system in the above-mentioned embodiment, and can solve the technical problem of low efficiency in processing audio data in the traditional solution. Compared with the prior art, the beneficial effects of the control device of the audio processing system provided by the present application are the same as those of the control method of the audio processing system provided by the above-mentioned embodiment, and other technical features in the control device of the audio processing system are the same as the features disclosed in the method of the previous embodiment, which will not be elaborated here.

[0130] It should be understood that each part disclosed in the present application can be implemented by hardware, software, firmware or a combination thereof. In the description of the above embodiments, specific features, structures, materials or characteristics can be combined in a suitable manner in any one or more embodiments or examples.

[0131] As mentioned above, the above are only specific embodiments of the present application, but the protection scope of the present application is not limited thereto. Any person skilled in the art can easily think of changes or substitutions within the technical scope disclosed in the present application, and all should be covered by the protection scope of the present application. Therefore, the protection scope of the present application should be subject to the protection scope of the claims.

[0132] The present application provides a computer-readable storage medium having computer-readable program instructions (i.e., computer programs) stored thereon, and the computer-readable program instructions are used to execute the control method of the audio processing system in the above-mentioned embodiment.

[0133] The computer-readable storage medium provided by the present application can be, for example, a USB flash drive, but is not limited to electrical, magnetic, optical, electromagnetic, infrared, or semiconductor systems, devices, or components, or any combination of the above. More specific examples of the computer-readable storage medium may include, but are not limited to: electrical connections with one or more wires, portable computer disks, hard disks, random access memory (RAM), read-only memory (ROM), erasable programmable read-only memory (EPROM), or flash memory, optical fibers, portable compact disk read-only memory (CD-ROM), optical storage devices, magnetic storage devices, or any suitable combination of the above. In this embodiment, the computer-readable storage medium can be any tangible medium that contains or stores a program, and this program can be used by or in combination with an instruction execution system, device, or component. The program code contained on the computer-readable storage medium can be transmitted by any suitable medium, including but not limited to: wires, optical cables, radio frequency (RF), etc., or any suitable combination of the above.

[0134] The above computer-readable storage medium can be included in the control device of the audio processing system; or it can exist independently without being assembled into the control device of the audio processing system.

[0135] The above computer-readable storage medium carries one or more programs. When the one or more programs are executed by the control device of the audio processing system, the control device of the audio processing system is enabled to: obtain the environmental information and device status data collected by the sensor; input the environmental information and the device status data into the clock compensation module to generate corresponding compensation parameters, and perform a compensation action on the audio clock based on the compensation parameters; determine the source sampling rate corresponding to at least one audio source according to the compensated audio clock; detect the source sampling rate through the sampling rate conversion module, and when the source sampling rate is different from the preset sampling rate, adjust the source sampling rate based on the preset sampling rate.

[0136] Computer program code for performing the operations of this application can be written in one or more programming languages or combinations thereof. The above-mentioned programming languages include object-oriented programming languages such as Java, Smalltalk, C++, and also include conventional procedural programming languages such as the "C" language or similar programming languages. The program code can be executed entirely on the user's computer, partially on the user's computer, executed as an independent software package, partially on the user's computer and partially on a remote computer, or entirely on a remote computer or server. In the case of a remote computer, the remote computer can be connected to the user's computer through any kind of network, including a local area network (LAN: Local Area Network) or a wide area network (WAN: Wide Area Network), or it can be connected to an external computer (for example, by connecting through an Internet service provider using the Internet).

[0137] The flowcharts and block diagrams in the accompanying drawings illustrate the possible architectures, functions, and operations of systems, methods, and computer program products according to various embodiments of this application. In this regard, each block in the flowchart or block diagram may represent a module, a program segment, or a part of code that contains one or more executable instructions for implementing a specified logical function. It should also be noted that in some alternative implementations, the functions marked in the blocks may occur in a different order than marked in the accompanying drawings. For example, two consecutive blocks shown may actually be executed substantially in parallel, and they may sometimes be executed in the 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 by a dedicated hardware-based system for performing the specified functions or operations, or can be implemented by a combination of dedicated hardware and computer instructions.

[0138] The modules described in the embodiments of this application can be implemented in software or in hardware. Among them, the name of the module does not constitute a limitation on the unit itself in some cases.

[0139] The readable storage medium provided in this application is a computer-readable storage medium. The computer-readable storage medium stores computer-readable program instructions (i.e., computer programs) for performing the control method of the above audio processing system, and can solve the technical problem of low efficiency when processing audio data in the traditional solution. Compared with the prior art, the beneficial effects of the computer-readable storage medium provided in this application are the same as those of the control method of the audio processing system provided in the above embodiments, and will not be elaborated here.

[0140] An embodiment of the present application provides a computer program product, including a computer program, which implements the steps of the control method of the audio processing system as described above when executed by a processor.

[0141] The computer program product provided by the present application can solve the technical problem of low efficiency in processing audio data in the traditional solution. Compared with the prior art, the beneficial effects of the computer program product provided by the embodiment of the present application are the same as those of the control method of the audio processing system provided by the above embodiment, and will not be elaborated here.

[0142] The above are only the preferred embodiments of the present application, and do not limit the patent scope of the present application. Any equivalent structure or equivalent process transformation made by using the content of the specification and drawings of the present application, or directly or indirectly applied in other related technical fields, shall be equally included in the patent scope of the present application.

Claims

1. A control method for an audio processing system, characterized in that, Applied to an audio processing system, the audio processing system includes a clock compensation module and a sampling rate conversion module, and the control method of the audio processing system includes the following steps: Obtain the environmental information and device status data collected by the sensor. The environmental information includes environmental temperature, humidity, and air pressure, and the device status data includes device load and device running time; Input the environmental information and the device status data into the clock compensation module to generate corresponding compensation parameters, and perform a compensation action on the audio clock based on the compensation parameters; Determine the source sampling rate corresponding to at least one audio source according to the compensated audio clock; Detect the source sampling rate through the sampling rate conversion module. When the source sampling rate is different from the preset sampling rate, adjust the source sampling rate based on the preset sampling rate; Collect the audio delay, buffer size, and remaining data volume in the audio processing process based on the audio delay control module; Input the audio delay, buffer size, and remaining data volume into the predictive read-write scheduling model, and determine the prediction result output by the predictive read-write scheduling model; Generate an audio processing strategy according to the prediction result, and optimize the audio processing process based on the audio processing strategy.

2. The method according to claim 1, wherein Before the step of obtaining the environmental information and device status data collected by the sensor, it further includes: Obtain the initial timestamp recorded based on the system clock through the clock compensation module, and determine the time difference according to the current clock value; Determine the audio source sampling rate corresponding to the initial timestamp, and determine the number of audio samples corresponding to the audio source sampling rate according to the time difference; Obtain the actual number of samples corresponding to the buffer, compare the actual number of samples with the number of audio samples, and if they are different, perform the corresponding compensation action.

3. The method according to claim 2, characterized in that, The step of performing the corresponding compensation action includes: Determine the ratio of the audio source sampling rate to the actual sampling rate corresponding to the actual number of samples; Determine the target read speed according to the ratio and the number of samples corresponding to the read speed; Adjust the target read speed to the current read speed.

4. The method according to claim 1, characterized in that, The step of detecting the source sampling rate through the sampling rate conversion module and adjusting the source sampling rate based on the preset sampling rate when the source sampling rate is different from the preset sampling rate includes: Obtain the preset sampling rate; Compare the source sampling rate and the preset sampling rate through the sampling rate conversion module; When the source sampling rate is less than the preset sampling rate, determine the proportional relationship according to the source sampling rate and the preset sampling rate; Determine the sample points to be inserted into the source sampling rate according to the proportional relationship, and insert the sample points into the source sampling rate based on the preset interpolation type.

5. The method according to claim 4, wherein After the step of determining the sample points to be inserted into the source sampling rate according to the proportional relationship and inserting the sample points into the source sampling rate based on the preset interpolation type, it further includes: When the source sampling rate is greater than the preset sampling rate, determine the proportional relationship between the source sampling rate and the preset sampling rate; Downsample the source sampling rate according to the proportional relationship.

6. The method according to claim 1, characterized in that, After the step of detecting the source sampling rate by the sampling rate conversion module and adjusting the source sampling rate based on the preset sampling rate when the source sampling rate is different from the preset sampling rate, the method further includes: Detecting system resource information by an audio delay control module; When the system resource information is greater than or equal to a preset threshold, setting the read / write speed of audio data to a first preset speed to reduce the read / write beats of the audio data; When the system resource information is less than the preset threshold, setting the read / write speed of the audio data to a second preset speed.

7. The method according to claim 6, wherein After the step of detecting system resource information by the audio delay control module, the method further includes: Obtaining a to-be-processed audio task in a to-be-processed queue and determining the priority of the to-be-processed audio task; When the system resource information is greater than or equal to a preset threshold, determining a target audio task with a priority higher than a preset priority among the to-be-processed audio tasks; Placing the target audio task at a priority processing position in the to-be-processed queue.

8. A control device for an audio processing system, characterized in that, The control device of the audio processing system includes: a memory, a processor, and a computer program stored on the memory and executable on the processor, where the computer program is configured to implement the steps of the control method of the audio processing system according to any one of claims 1 to 7.

9. A storage medium, characterized in that, The storage medium is a computer-readable storage medium, and a computer program is stored on the computer-readable storage medium. When the computer program is executed by a processor, the steps of the control method of the audio processing system according to any one of claims 1 to 7 are implemented.

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