CN115278456B - A kind of sound equipment and audio signal processing method - Google Patents

Abstract

The embodiment of the application provides sound equipment and an audio signal processing method, relates to the technical field of audio signal processing, and solves the problems of insufficient tone quality definition and poor sound effect layering effect in the related technology. The sound equipment comprises a sampling rate converter, a first-order harmonic generator, a second-order harmonic generator, a loudspeaker and a controller; the controller is configured to: controlling a sampling rate converter to convert a first audio signal of a first sampling frequency into a second audio signal of a second sampling frequency in an audio file in response to a play operation performed by a user on the audio file; controlling a first-stage harmonic generator to generate first harmonic information according to the first audio signal, and generating a third audio signal according to the first harmonic information and the second audio signal; controlling a second harmonic generator to generate second harmonic information according to the third audio signal, and generating a fourth audio signal according to the second harmonic information and the third audio signal; the control speaker outputs a target audio signal based on the fourth audio signal.

Description

A kind of sound equipment and audio signal processing method

Technical Field

The present application relates to the technical field of audio signal processing, and in particular to an audio device and an audio signal processing method.

Background Art

At present, in order to meet the reasonable utilization of audio resources, an audio digital information compression method is adopted to compress audio resources into audio files, and then the content of the audio resources is played by decompressing the audio files. However, in the above compression process, the mid-high frequency signals in the audio resources are filtered, and some useful mid-high frequency signals are lost, resulting in insufficient sound quality clarity and poor sound effect layering when the audio files are directly decompressed to play the content of the audio resources.

Summary of the invention

The embodiments of the present application provide an audio device and an audio signal processing method to at least solve the problems of insufficient sound clarity and poor sound effect layering in the related art.

In a first aspect, an audio device is provided, comprising a sampling rate converter, a primary harmonic generator, a secondary harmonic generator, a speaker and a controller; the controller is connected to the sampling rate converter, the primary harmonic generator, the secondary harmonic generator and the speaker; the controller is configured to: in response to a user's playback operation performed on an audio file, control the sampling rate converter to convert a first audio signal of a first sampling frequency in the audio file into a second audio signal of a second sampling frequency, the first sampling frequency being less than the second sampling frequency, and the sampling amplitude of the first audio signal being less than the sampling amplitude of the second audio signal; control the primary harmonic generator to generate first harmonic information according to the first audio signal, and to generate a third audio signal according to the first harmonic information and the second audio signal; control the secondary harmonic generator to generate second harmonic information according to the third audio signal, and to generate a fourth audio signal according to the second harmonic information and the third audio signal; control the speaker to output a target audio signal according to the fourth audio signal, wherein the frequency of the first harmonic information is less than the frequency of the second harmonic information.

The technical solution provided by the embodiment of the present application brings at least the following beneficial effects: when performing a playback operation on a compressed audio file, the first audio signal in the audio file is first converted into a second audio signal through a sampling rate converter of an audio device, wherein the second sampling frequency of the second audio signal is higher than the first sampling frequency of the first audio signal, thereby improving the resolution of the audio signal, and the sampling amplitude of the second audio signal is greater than the sampling amplitude of the second audio signal, so as to prepare the time domain space for the subsequent generation of harmonic signal insertion. Then, the first harmonic information is generated according to the first harmonic generator, and the first harmonic signal compensation is performed on the second audio signal to generate a third audio signal, so as to restore or compensate for the harmonic signal lost during the compression process of the audio file, and ensure that the audio output of the audio device is clearer. Then, according to the second harmonic information generated by the second harmonic generator, the third audio signal is compensated for the harmonic signal, that is, the second audio signal is compensated for the second harmonic signal to generate a fourth audio signal, so as to ensure that the audio output of the audio device has a more layered and spatial sense. Finally, the target audio signal is output according to the fourth audio signal to ensure the audio effect of the audio device. Through the above method, on the one hand, the harmonic signal of the frequency of the first harmonic information lost in the compression process of the audio file is repaired to restore the sound characteristics lost in the audio file, thereby ensuring that the audio output by the audio device retains the sound characteristics of the original audio to a greater extent, reducing the distortion of the audio output by the audio device and improving the clarity of the output audio. On the other hand, the harmonic signal of the harmonic signal of the frequency of the second harmonic information lost in the compressed audio file is compensated to improve the layering and spatial sense of the audio output by the audio device.

In a possible implementation, the first harmonic information is used to generate medium and high frequency harmonic signals, and the second harmonic information is used to generate ultra-high frequency harmonic signals. Based on this, the medium and high frequency harmonic signals lost in the compression process of the audio file are repaired according to the first harmonic information to restore the sound characteristics lost in the audio file, thereby ensuring that the audio output by the audio device retains the sound characteristics of the original audio to a greater extent, reducing the distortion of the audio output by the audio device, and improving the clarity of the output audio. Furthermore, according to the second harmonic information, the ultra-high frequency audio signal lost in the compressed audio file is compensated by the ultra-high frequency harmonic signal to improve the layering and spatial sense of the audio output by the audio device.

In some embodiments, the first audio signal is an audio signal in a two-channel pulse code modulation format. Before controlling the sampling rate converter to convert the first audio signal of a first sampling frequency in the audio file into a second audio signal of a second sampling frequency, it also includes: when the audio file includes audio signals in other formats, converting the audio signals in other formats into an audio signal in a two-channel pulse code modulation format, and the other formats include formats other than the two-channel pulse code modulation format.

Based on this, the audio signal in the audio file is converted into a unified dual-channel pulse code modulation format, so that there is a unified multiple conversion relationship between the sampling frequency of the first audio signal and the frequency of the output target audio signal, that is, the sampling frequency of the audio signal is twice the frequency of the output target audio signal, so that the sampling rate converter can quickly convert the first audio signal into a second audio signal of a second sampling frequency, and at the same time, before the target audio signal is output, the audio signal can be better distributed to each channel. In addition, the dual-channel pulse code modulation format can be understood as a left and right two-channel pulse code modulation format, and other formats can be understood as non-two-channel formats and/or non-pulse code modulation formats, such as three-channel formats, four-channel formats, pulse density modulation formats, etc.

In some embodiments, a third audio signal is generated based on first harmonic information and a second audio signal, including: based on the first harmonic information, harmonic compensation is performed on an audio signal within a first frequency band in the second audio signal to obtain a third audio signal, the frequency of the first harmonic information belongs to the first frequency band, and the first frequency band is a frequency range in which the sampling frequency is less than the first sampling frequency.

In this embodiment, the first harmonic information is generated based on the first audio signal. According to the first harmonic information, the audio signal in the frequency range less than the first sampling frequency in the second audio signal is compensated for the mid-high frequency harmonic signal, so as to complete the mid-high frequency harmonic signal compensation for the first audio signal in the second audio signal, thereby repairing the mid-high frequency harmonic signal lost in the audio file compression process, so as to restore the sound characteristics lost in the audio file, thereby ensuring that the audio output by the audio device retains the sound characteristics of the original audio to a greater extent, reducing the distortion of the audio output by the audio device, and improving the clarity of the output audio.

In some embodiments, a fourth audio signal is generated based on the second harmonic information and the third audio signal, including: converting the frequencies of the corresponding audio signals in the second frequency band in the third audio signal into frequencies greater than or equal to the second sampling frequency; the second frequency band is the frequency range between the first sampling frequency and the second sampling frequency, and the frequency of the second harmonic information belongs to the second frequency band; based on the second harmonic information, harmonic compensation is performed on the audio signals in the third audio signal that are greater than or equal to the second sampling frequency to obtain the fourth audio signal.

In this embodiment, the frequencies of the audio signals in the third harmonic signal within the frequency range greater than or equal to the first sampling frequency and less than or equal to the second sampling frequency are first converted into sampling frequencies greater than or equal to the second sampling frequency, so that there is sufficient time domain space for subsequent insertion of the ultra-high frequency harmonic signal. The ultra-high frequency harmonic signal generated according to the second harmonic information is then compensated into the third audio signal.

In some embodiments, the audio device also includes: a variable resolution impulse response filter; the variable resolution impulse response filter is connected to the controller; before controlling the speaker to output the target audio signal according to the fourth audio signal, the controller is also configured to: control the variable resolution impulse response filter to divide the fourth audio signal into multiple audio signals of different frequency bands; according to the filtering parameters corresponding to each frequency band, filter the audio signals of each frequency band respectively to obtain a fifth audio signal, wherein the fifth audio signal includes an audio signal obtained by filtering the audio signals of multiple frequency bands; according to the fifth audio signal, generate a target audio signal of a preset frequency band; the preset frequency band is a frequency range lower than the first sampling frequency; the controller controls the speaker to output the target audio signal according to the fourth audio signal, and is configured to: control the speaker to output the target audio signal of the preset frequency band.

It is understandable that audio signals of different frequency bands refer to audio signals of different sampling frequency ranges. And the filter parameters corresponding to each frequency band refer to the filter parameters corresponding to each frequency band being interrelated and having mutual influence. That is, the filter parameters corresponding to different frequency bands influence each other.

In this embodiment, according to the variable resolution impulse response filter, the audio signals of multiple different frequency bands in the fourth audio signal are first subjected to segmented filtering processing. That is, for the audio signals of multiple different frequency bands, filtering processing is performed using the filtering parameters corresponding to the audio signals of each frequency band. Then, the audio signals of each frequency band after filtering processing are used as the fifth audio signal. Based on this, the audio signals of different sampling frequencies in the fourth audio signal are subjected to separate filtering processing, and different resolution requirements can be flexibly integrated to adjust the fifth harmonic signal so that the output target audio signal meets the multi-resolution requirements, avoiding the problem of poor flexibility of the audio equipment caused by uniform filtering processing of the audio signal without frequency band, which can only meet a single resolution requirement, thereby ensuring the flexibility and accuracy of the audio equipment.

In some embodiments, the phase of the speaker is the same as the phase of the variable resolution impulse response filter. Based on this, the phase of the variable resolution impulse response filter is kept at the same phase as the phase of the speaker, so that the audio signal processed by the variable resolution impulse response filter is input to the speaker and processed by the speaker, and the output target audio signal is more balanced, thereby ensuring that the sound quality of the audio output by the audio device is more balanced.

In some embodiments, before controlling the sampling rate converter to convert the first audio signal of the first sampling frequency in the audio file into the second audio signal of the second sampling frequency, the method further includes: parsing the audio file to obtain the first audio signal. Based on this, the audio file is parsed into an audio signal to provide an audio signal source for the subsequent audio signal processing link.

In a second aspect, a method for processing an audio signal is provided, the method comprising: in response to a playback operation performed by a user on an audio file, converting a first audio signal of a first sampling frequency in the audio file into a second audio signal of a second sampling frequency, the first sampling frequency being less than the second sampling frequency, and the sampling amplitude of the first audio signal being less than the sampling amplitude of the second audio signal; generating first harmonic information according to the first audio signal, and generating a third audio signal according to the first harmonic information and the second audio signal; generating second harmonic information according to the third audio signal, and generating a fourth audio signal according to the second harmonic information and the third audio signal; controlling a speaker to output a target audio signal according to the fourth audio signal, wherein the frequency of the first harmonic information is less than the frequency of the second harmonic information.

In some embodiments, the first audio signal is an audio signal in a dual-channel pulse code modulation format, and before converting the first audio signal of a first sampling frequency in the audio file into a second audio signal of a second sampling frequency, the method further includes:

In the case that the audio file includes audio signals in other formats, the audio signals in other formats are converted into audio signals in two-channel pulse code modulation format, and the other formats include formats other than the two-channel pulse code modulation format.

In some embodiments, a third audio signal is generated based on first harmonic information and a second audio signal, including: based on the first harmonic information, harmonic compensation is performed on an audio signal within a first frequency band in the second audio signal to obtain a third audio signal, the frequency of the first harmonic information belongs to the first frequency band, and the first frequency band is a frequency range in which the sampling frequency is less than the first sampling frequency.

In some embodiments, a fourth audio signal is generated based on the second harmonic information and the third audio signal, including: converting the sampling frequencies of the audio signals in the second frequency band in the third audio signal into the second sampling frequency; the second frequency band is the frequency range between the first sampling frequency and the second sampling frequency, and the frequency of the second harmonic information belongs to the second frequency band; and based on the second harmonic information, performing harmonic compensation on the audio signal of the second sampling frequency in the third audio signal to obtain the fourth audio signal.

In some embodiments, before outputting the target audio signal according to the fourth audio signal, the method also includes: dividing the fourth audio signal into audio signals of multiple frequency bands; filtering the audio signals of the multiple frequency bands according to the filtering parameters corresponding to each frequency band to obtain a fifth audio signal, wherein the fifth audio signal includes an audio signal obtained by filtering the audio signals of the multiple frequency bands; generating a target audio signal of a preset frequency band according to the fifth audio signal; the preset frequency band is a frequency range lower than the first sampling frequency; outputting the target audio signal according to the fourth audio signal includes: outputting the target audio signal lower than the preset frequency band.

In a third aspect, an embodiment of the present application provides a computer-readable storage medium, in which instructions are stored. When the instructions are executed on any of the above-mentioned devices, the device executes any of the above-mentioned audio signal processing methods of the audio equipment.

In a fourth aspect, an embodiment of the present application provides a chip, comprising: a processor and a memory; the memory is used to store computer execution instructions, the processor is connected to the memory, and when the chip is running, the processor executes the computer execution instructions stored in the memory so that the chip executes any of the above-mentioned audio signal processing methods of the audio device.

In a fifth aspect, an embodiment of the present application provides a computer program product comprising instructions, which, when executed on any of the above-mentioned devices, enables the device to execute the audio signal processing method of any of the above-mentioned audio devices.

In the embodiments of the present application, the names of the components of the above-mentioned device do not limit the device itself. In actual implementation, these components may appear with other names. As long as the functions of each component are similar to the embodiments of the present application, they fall within the scope of the claims of the present application and their equivalent technologies.

In addition, the technical effects brought about by any design method in the second to fifth aspects can refer to the technical effects brought about by the different design methods in the above-mentioned first aspect, and will not be repeated here.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG1 is a circuit system architecture diagram of an audio device provided in an embodiment of the present application;

FIG2 is a schematic diagram of a usage scenario of an audio device provided in an embodiment of the present application;

FIG3 is a schematic diagram of a usage scenario of another audio device provided in an embodiment of the present application;

FIG4 is a flow chart of an audio signal processing method of an audio device provided in an embodiment of the present application;

FIG5 is a schematic diagram of a sampling frequency conversion process provided by an embodiment of the present application;

FIG6 is a schematic diagram of harmonic compensation of a primary harmonic generator of an audio device provided in an embodiment of the present application;

FIG7 is a flowchart of another audio signal processing method for an audio device provided in an embodiment of the present application;

FIG8 is a flowchart of another audio signal processing method for an audio device provided in an embodiment of the present application;

FIG9 is a flowchart of another method for processing audio signals of an audio device provided in an embodiment of the present application;

FIG10 is a schematic diagram of a variable resolution impulse response filter filtering process of an audio device provided by an embodiment of the present application;

FIG11 is a schematic diagram of a usage scenario of another audio device provided in an embodiment of the present application;

FIG12 is a schematic diagram of an audio signal processing process of an audio device provided in an embodiment of the present application;

FIG. 13 is a schematic diagram of the hardware structure of a controller provided in an embodiment of the present application.

DETAILED DESCRIPTION

The following will be combined with the drawings in the embodiments of the present application to clearly and completely describe the technical solutions in the embodiments of the present application. Obviously, the described embodiments are only part of the embodiments of the present application, not all of the embodiments. Based on the embodiments in the present application, all other embodiments obtained by ordinary technicians in this field without creative work are within the scope of protection of this application.

The terms "first" and "second" are used for descriptive purposes only and should not be understood as indicating or implying relative importance or implicitly indicating the number of the indicated technical features. Thus, a feature defined as "first" or "second" may explicitly or implicitly include one or more of the features. In the description of this application, unless otherwise specified, "plurality" means two or more.

In the description of this application, it should be noted that, unless otherwise clearly specified and limited, the terms "connected" and "connection" should be understood in a broad sense, for example, it can be a fixed connection, a detachable connection, or an integral connection. For ordinary technicians in this field, the specific meanings of the above terms in this application can be understood according to specific circumstances. In addition, when describing a pipeline, the "connected" and "connection" used in this application have the meaning of conduction. The specific meaning needs to be understood in conjunction with the context.

In the embodiments of the present application, words such as "exemplary" or "for example" are used to indicate examples, illustrations or descriptions. Any embodiment or design described as "exemplary" or "for example" in the embodiments of the present application should not be interpreted as being more preferred or more advantageous than other embodiments or designs. Specifically, the use of words such as "exemplary" or "for example" is intended to present related concepts in a specific way.

At present, in order to meet the reasonable utilization of audio resources, an audio digital information compression method is adopted to compress audio resources into audio files, and then the content of the audio resources is played by decompressing the audio files. However, in the above compression process, the mid-high frequency signals in the audio resources are filtered, and some useful mid-high frequency signals are lost, resulting in insufficient sound quality clarity and poor sound effect layering when the audio files are directly decompressed to play the content of the audio resources.

In view of this, an embodiment of the present application provides an audio device, which includes a sampling rate converter, a first-level harmonic generator, a second-level harmonic generator, a speaker and a controller, wherein the controller is connected to the sampling rate converter, the first-level harmonic generator, the second-level harmonic generator and the speaker, and is used to control the sampling rate converter, the first-level harmonic generator, and the second-level harmonic generator to process the audio signal to be processed, and control the speaker to output the processed audio signal. When performing a playback operation on a compressed audio file, the first audio signal in the audio file is first converted into a second audio signal by the sampling rate converter of the audio device, wherein the second sampling frequency of the second audio signal is higher than the first sampling frequency of the first audio signal, thereby improving the resolution of the audio signal, and the sampling amplitude of the second audio signal is greater than the sampling amplitude of the second audio signal, so as to prepare the time domain space for the subsequent generated harmonic signal insertion. Then, the first harmonic information is generated according to the first-level harmonic generator, and the first harmonic signal compensation is performed on the second audio signal to generate a third audio signal, so as to restore or compensate the harmonic signal lost during the compression process of the audio file, and ensure that the audio output of the audio device is clearer. Then, according to the second harmonic information generated by the secondary harmonic generator, the third audio signal is compensated for the harmonic signal, that is, the second audio signal is compensated for the second harmonic signal to generate a fourth audio signal, so as to ensure that the audio output by the audio device has a better sense of hierarchy and space. Finally, the target audio signal is output according to the fourth audio signal to ensure the audio effect of the audio device. Through the above method, on the one hand, the harmonic signal of the frequency of the first harmonic information lost in the compression process of the audio file is repaired to restore the sound characteristics lost in the audio file, thereby ensuring that the audio output by the audio device retains the sound characteristics of the original audio to a greater extent, reducing the distortion of the audio output by the audio device and improving the clarity of the output audio. On the other hand, the harmonic signal of the frequency of the second harmonic information lost in the compressed audio file is compensated for the harmonic signal to improve the sense of hierarchy and space of the audio output by the audio device.

Through this implementation, on the one hand, the mid-high frequency harmonic signals lost in the compression process of the audio file are repaired to restore the sound characteristics lost in the audio file, thereby ensuring that the audio output by the audio device retains the sound characteristics of the original audio to a greater extent, reducing the distortion of the audio output by the audio device and improving the clarity of the output audio. On the other hand, the ultra-high frequency audio signal lost in the compressed audio file is compensated by the ultra-high frequency harmonic signal to improve the layering and spatial sense of the audio output by the audio device.

In the embodiments of the present application, the audio device can be understood as an electronic device with an audio resource playback function. Exemplarily, the audio device can be an embedded audio device, for example, an audio output device of an electronic device, and the electronic device can be a mobile phone, a computer, or a server, etc. In another exemplary embodiment, the audio device can also be an independent audio device, such as a wired speaker, a wireless Bluetooth speaker, a wireless network speaker, etc. Therefore, the present application does not specifically limit the specific form of the audio device, and it can be specifically set according to specific needs.

Referring to FIG1 , there is shown an exemplary circuit system architecture diagram of an audio device 10. The audio device 10 comprises: a sampling rate converter 101, a primary harmonic generator 102, a secondary harmonic generator 103, a speaker 104 and a controller 105. The controller 105 is connected to the sampling rate converter 101, the primary harmonic generator 102, the secondary harmonic generator 103 and the speaker 104. The sampling rate converter 101 is used to convert the sampling frequency of the audio signal in the audio file; the first harmonic information is used to generate a harmonic signal of a certain frequency band, such as generating a medium-high frequency harmonic signal, and the second harmonic information is used to generate a harmonic signal of another frequency band, such as generating an ultra-high frequency harmonic signal. The speaker 104 is used to output the audio signal.

In some embodiments, the format of the audio file can be MP3 format or AAC format. The present application does not specifically limit the specific format of the audio file.

In some embodiments, the first harmonic generator 102 may be an audio exciter that generates a harmonic signal proportional to an audio signal input to the audio exciter. Exemplarily, the frequency of the harmonic signal generated by the first harmonic generator 102 is a second-order exponential multiple of the frequency of the audio signal input to the first harmonic generator 102. For example, the harmonic signal generated by the first harmonic generator 102 is 2 times, 4 times (2 to the power of 2), or 8 times (2 to the power of 3), etc., the frequency of the audio signal input to the first harmonic generator 102.

In some embodiments, the audio device 10 further includes: a parser 106, wherein the parser 106 is configured to parse the audio file to obtain an audio signal to be processed (ie, the first audio signal described below).

In some embodiments, the audio device 10 further includes: a filter.

Optionally, the filter may be a variable resolution impulse response filter 108, wherein the variable resolution impulse response filter 108 is connected to the controller 105. The variable resolution impulse response filter 108 determines the filter parameters according to the frequency, gain, quality factor and the influence factor of each frequency band. During the operation of the variable resolution impulse response filter 108, each frequency band can communicate with each other and is not relatively independent, so that the influence of each frequency band can be considered to adjust the audio signal of each frequency band.

In some embodiments, the variable-resolution impulse response filter 108 is also referred to as a VIR filter (variable-resolution impulse response). When the sampling frequency increases, the MIPS value (million instructions per second, an indicator of the computing speed of the number of machine language instructions processed per second) of the VIR filter increases proportionally. However, it does not pointlessly place a portion of the audio signals within the sampling frequency range within the ultrasonic range like an ordinary filter. Therefore, the Hi-Res (High Resolution Audio) effect of the audio processed by the VIR filter is better. In addition, when the VIR filter is used interactively with the processor, since the MIPS value of the VIR filter changes with the sampling frequency, the delay when the VIR filter is used interactively with the processor is greatly reduced.

In some embodiments, the audio device 10 further includes a display 107. The display 107 can be used to display a control panel or other image information of the audio device 10. For example, the audio device can display the current audio file processing process or the waveform of the audio signal of the audio device 10 through the display 107.

In addition, the display 107 may be a liquid crystal display 107 or an organic light-emitting diode (OLED) display 107. The specific type, size, and resolution of the display 107 are not limited, and those skilled in the art will appreciate that the display 107 may be modified in performance and configuration as required.

Exemplarily, as shown in FIG. 2 , the user may operate the audio device 10 via a control panel displayed on the display 107 .

In some embodiments, the controller 105 refers to a device that can generate an operation control signal according to the instruction operation code and the timing signal to instruct the audio device 10 to execute the control instruction. Exemplarily, the controller 105 can be a central processing unit (CPU), a general-purpose processor network processor (NP), a digital signal processor (DSP), a programmable logic device (PLD), a microprocessor, a microcontroller or any combination thereof. The controller 105 can also be other devices with processing functions, such as circuits, devices or software modules, and the embodiments of the present application do not impose any restrictions on this.

In some embodiments, the audio device 10 further includes a communication device 109, which is a component for communicating with an external device or an external server according to various communication protocol types. For example, the communication device 109 may include at least one of a Wi-Fi chip, a Bluetooth communication protocol chip, a wired Ethernet communication protocol chip, other network communication protocol chips or a near field communication protocol chip, and an infrared receiver.

In some embodiments, the audio device 10 can transmit control signals and data signals to a server of a terminal device (e.g., a mobile phone, a tablet computer, a wearable mobile device, etc.) used by a user through a communication device 109. For example, a user issues a play instruction for a certain audio file through a mobile phone, and the audio device 10 receives the instruction through the communication device 109. In response to the user's instruction to play a certain audio file, the controller 105 of the audio device 10 plays the audio file.

Exemplarily, as shown in FIG3 , the user can operate the audio device 10 through the control device 112. The control device can be a mobile terminal or a remote controller, and the communication between the remote controller and the audio device 10 includes infrared protocol communication or Bluetooth protocol communication, and other short-range communication methods, etc., and the audio device 10 is controlled wirelessly or by other wired methods. Among them, the wireless method can be a direct connection or an indirect connection, and can be routed or not. The user can input user instructions through buttons on the remote controller, voice input, control panel input, etc. to control the audio device 10. For example, the user can input corresponding control instructions through the volume plus and minus keys, channel control keys, up/down/left/right movement keys, voice input keys, menu keys, power on/off keys, etc. on the remote controller to realize the function of controlling the audio device 10.

In some embodiments, the audio device 10 further includes a human-computer interaction device 110 for realizing interaction between the user and the audio device 10. The human-computer interaction device 110 may include one or more of a physical button, a touch display panel, or a voice recognition device. For example, the user can start the audio device 10 through the human-computer interaction device 110 to start working, or can set the audio device 10 to play an audio file through the human-computer interaction device 110.

In some embodiments, the audio device 10 further includes a power supply 111 for providing power supply support to the audio device 10 using power input from an external power source under the control of the controller 105 .

Based on the above-mentioned audio device 10, as shown in FIG4 , an embodiment of the present application provides an audio signal processing method, the method comprising the following steps:

Step S101 : in response to a user's play operation on an audio file, converting a first audio signal of a first sampling frequency in the audio file into a second audio signal of a second sampling frequency.

The first sampling frequency is smaller than the second sampling frequency, and the sampling amplitude of the first audio signal is smaller than the sampling amplitude of the second audio signal.

In some implementations, the sampling amplitude is also referred to as sampling depth, the unit of sampling depth is bit, and the unit of sampling frequency is Hz. The first audio signal is usually a 16-bit 32k Hz audio signal, a 16-bit 44.1k Hz audio signal, or a 16-bit 48k Hz audio signal.

Optionally, the 16-bit 48 kHz first audio signal is converted into a 32-bit 96 kHz second audio signal.

It should be noted that, as shown in FIG. 5 , hollow points are added to the audio signal processed by the sampling rate converter, that is, relative to the first audio signal, only the sampling frequency and sampling amplitude of the second audio signal are changed, and the second audio signal includes the first audio signal.

An audio file refers to an audio file in various formats compressed from an original audio signal. Exemplarily, the audio file may be in MP3 format or AAC format. For example, the audio file may be an audio resource included in an application installed on a mobile terminal. The mobile terminal may be a mobile phone or a computer, and an application may be a music player application, and the audio file is an audio resource of a certain song. Furthermore, when the music player application is activated, an audio file identifier (such as the name of a certain song) may be displayed on an application interface displayed on the mobile device, and after the audio file identifier is triggered, the audio resource of a certain song may be output.

For another example, the audio file may be an audio file including audio resources stored in an electronic device, and the electronic device may be a mobile terminal device, or an audio playback device (such as a speaker device) used solely for amplifying audio files.

Based on this, when the user needs to obtain the audio resource of a certain audio file, the play operation of the audio file will be performed. The play operation can be performed directly on the audio file identifier displayed on the electronic device, such as a single-click operation, a double-click operation, a continuous click operation or a sliding operation. After the controller receives the play operation instruction of the user, the first audio signal of the audio file is first obtained. And in order to improve the resolution of the audio signal in the subsequent process, the first audio signal will be converted into a second audio signal with a higher sampling frequency (a second sampling frequency). Among them, the conversion of the first audio signal into the second audio signal can be implemented by a sampling converter of the audio equipment.

Exemplarily, in some embodiments, the audio device further includes a parser for parsing the audio file into an audio signal. In this way, the first audio signal can be obtained by the following implementation: before the controller controls the sampling rate converter to convert the first audio signal of the first sampling frequency in the audio file into the second audio signal of the second sampling frequency, the audio file is parsed by controlling the parser to obtain the first audio signal. The parser is connected to the controller. Based on this implementation, the audio file is parsed into an audio signal, providing an audio signal source for the subsequent audio signal processing link.

Through the above S101, the first audio signal is converted into a second audio signal with a higher sampling frequency, thereby improving the resolution of the audio signal. At the same time, the sampling amplitude of the second audio signal is greater than the sampling amplitude of the second audio signal, and the time domain space is prepared for the insertion of the subsequently generated harmonic signal.

Step S102: generating first harmonic information according to the first audio signal, and generating a third audio signal according to the first harmonic information and the second audio signal.

It can be understood that the first harmonic information is first generated according to the original audio signal of the audio file, that is, the first audio signal, and the first harmonic information includes the harmonic signal missing from the first audio signal, that is, the first harmonic signal. Then, the first harmonic signal is superimposed on the first audio signal included in the second audio signal, thereby generating the third audio signal.

Exemplarily, the controller controls the primary harmonic generator to generate the first harmonic information of the middle and high frequencies according to the first audio signal, and performs the middle and high frequency harmonic signal compensation on the first audio signal in the second audio signal to generate the third audio signal, wherein the waveform effects of the first audio signal and the third audio signal are shown in FIG6. It can be seen that the middle and high frequency harmonic information lost in the audio file compression process is restored or compensated in this way to ensure the clarity of the audio output by the audio device.

Step S103: generating second harmonic information according to the third audio signal, and generating a fourth audio signal according to the second harmonic information and the third audio signal.

Optionally, the controller controls the secondary harmonic generator to generate ultra-high frequency second harmonic information according to the third harmonic signal, performs ultra-high frequency harmonic signal compensation on the third audio signal, and generates a fourth audio signal.

In this way, through this implementation step, the audio output by the audio equipment is ensured to have a better sense of layering and space.

Step S104: output a target audio signal according to the fourth audio signal.

Optionally, the speaker is controlled to output a target audio signal according to the fourth audio signal to ensure that the audio output by the audio device has a greater sense of hierarchy and space.

The technical solution shown in FIG4 brings at least the following beneficial effects: on the one hand, the mid-high frequency harmonic signals lost in the compression process of the audio file are repaired to restore the sound characteristics lost in the audio file, thereby ensuring that the audio output by the audio device retains the sound characteristics of the original audio to a greater extent, reducing the distortion of the audio output by the audio device and improving the clarity of the output audio. On the other hand, the ultra-high frequency audio signal lost in the compressed audio file is compensated by the ultra-high frequency harmonic signal to improve the layering and spatial sense of the audio output by the audio device.

In some embodiments, the first audio signal is an audio signal in a two-channel pulse code modulation format.

Optionally, before controlling the sampling rate converter to convert a first audio signal of a first sampling frequency in an audio file into a second audio signal of a second sampling frequency, the method further includes: when the audio file includes audio signals in other formats, converting the audio signals in other formats into audio signals in a two-channel pulse code modulation format, wherein the other formats include formats other than the two-channel pulse code modulation format.

Based on this, the audio signal in the audio file is converted into a unified dual-channel pulse code modulation format, so that there is a unified multiple conversion relationship between the sampling frequency of the audio signal and the frequency of the output target audio signal, that is, the sampling frequency of the audio signal is twice the frequency of the output target audio signal, so that the sampling rate converter can quickly convert the first audio signal into a second audio signal with a second sampling frequency, and at the same time, before the target audio signal is output, the audio signal can be better distributed to each channel. In addition, the dual-channel pulse code modulation format can be understood as a left and right two-channel pulse code modulation format, and other formats can be understood as non-two-channel formats and/or non-pulse code modulation formats, such as three-channel formats, four-channel formats, pulse density modulation formats, etc.

In some embodiments, in combination with FIG. 4 , as shown in FIG. 7 , the above step S102 may be specifically implemented as the following steps:

Step S102A: perform harmonic compensation on the audio signal in the first frequency band of the second audio signal according to the first harmonic information to obtain a third audio signal.

It should be noted that the first harmonic information is used to generate medium and high frequency harmonic signals, and the medium and high frequency harmonic signals are inserted into the first audio signal included in the second audio signal to complete the medium and high frequency harmonic signal compensation.

The frequency of the first harmonic information belongs to a first frequency band, and the first frequency band is a frequency range in which the sampling frequency is less than the first sampling frequency.

In this embodiment, the first harmonic information is generated based on the first audio signal. According to the first harmonic information, the audio signal in the frequency range less than the first sampling frequency in the second audio signal is compensated for the mid-high frequency harmonic signal, so as to complete the mid-high frequency harmonic signal compensation for the first audio signal in the second audio signal, thereby realizing the repair of the mid-high frequency harmonic signal lost in the compression process of the audio file, so as to restore the sound characteristics lost in the audio file, thereby ensuring that the audio output by the audio device retains the sound characteristics of the original audio to a greater extent, reducing the distortion of the audio output by the audio device, and improving the clarity of the output audio.

In some embodiments, in combination with FIG. 4 , as shown in FIG. 8 , the above step S103 may be specifically implemented as the following steps:

Step S103A: convert the frequencies of the audio signals corresponding to the second frequency band in the third audio signal into sampling frequencies greater than or equal to the second sampling frequency.

The second frequency band is a frequency range between the first sampling frequency and the second sampling frequency, and the frequency of the second harmonic information belongs to the second frequency band.

Step S103B: performing harmonic compensation on the audio signal with a frequency greater than or equal to the second sampling frequency in the third audio signal according to the second harmonic information to obtain a fourth audio signal.

It should be noted that the second harmonic information is used to generate an ultra-high frequency harmonic signal, and the ultra-high frequency harmonic signal is inserted into a frequency band outside the frequency band corresponding to the first audio signal to complete ultra-high quality harmonic signal compensation.

In this embodiment, the frequencies of the audio signals in the third harmonic signal within the frequency range greater than or equal to the first sampling frequency and less than or equal to the second sampling frequency are first converted into sampling frequencies greater than or equal to the second sampling frequency, so that there is sufficient time domain space for subsequent insertion of the ultra-high frequency harmonic signal. The ultra-high frequency harmonic signal generated according to the second harmonic information is then compensated into the third audio signal.

Based on this embodiment, in combination with FIG. 4 , as shown in FIG. 9 , the above step S104 can be specifically implemented as the following steps:

Step S104A: controlling the variable resolution impulse response filter to divide the fourth audio signal into a plurality of audio signals in different frequency bands.

Based on this step S104A, the VIR filter can operate at any sampling frequency (e.g., 192K Hz) set by the system. The VIR filter allocates all available parts of the audio signal to the lower frequency bands of the multiple different frequency bands corresponding to the preset low frequency band range (e.g., below 100 Hz) to maximize the resolution, and the higher frequency bands of the multiple different frequency bands corresponding to the preset high frequency band range (e.g., above 100 Hz) are also utilized. Therefore, even in a smaller range (e.g., within the range of 10 Hz to 20 Hz) within the preset low frequency band range, the VIR filter can form a steep cutoff filter to eliminate unwanted rumble.

Step S104B: filtering the audio signals of each frequency band according to the filtering parameters corresponding to each frequency band to obtain a fifth audio signal.

The fifth audio signal includes an audio signal obtained by filtering audio signals in multiple frequency bands.

In some implementations, the filter coefficients of the VIR filter are determined based on the acoustic power volume density frequency response of the speaker.

In other implementations, the filter coefficients of the VIR filter may also be converted based on the filter coefficients of other non-filters.

Step S104C: generating a target audio signal according to the fifth audio signal.

In some embodiments, the frequency band to which the frequency of the target audio signal belongs may be a preset frequency band set according to the filtering parameters, and the preset frequency band is usually a frequency range lower than the first sampling frequency.

Step S104D: controlling the speaker to output a target audio signal of a preset frequency band.

It is understandable that audio signals of different frequency bands refer to audio signals of different sampling frequency ranges. And the filter parameters corresponding to each frequency band refer to the filter parameters corresponding to each frequency band being interrelated and having mutual influence. That is, the filter parameters corresponding to different frequency bands influence each other.

In this embodiment, according to the variable resolution impulse response filter, the audio signals of multiple different frequency bands in the fourth audio signal are first subjected to segmented filtering processing. That is, for the audio signals of multiple different frequency bands, filtering processing is performed using the filtering parameters corresponding to the audio signals of each frequency band. The audio signals of each frequency band after filtering processing are then used as the fifth audio signal. Based on this, the audio signals of different sampling frequencies in the fourth audio signal are subjected to separate filtering processing, and different resolution requirements can be flexibly integrated to adjust the fifth harmonic signal, so that the output target audio signal can meet the multi-resolution requirements, avoiding the problem of poor flexibility of the audio equipment caused by uniform filtering processing of the audio signal without dividing the frequency bands, which can only meet a single resolution requirement, thereby ensuring the flexibility and accuracy of the audio equipment.

Based on the above embodiment, the phase of the loudspeaker is the same as the phase of the variable resolution impulse response filter.

The loudspeaker is essentially a minimum phase device, so the variable resolution impulse response filter can also be a type of minimum phase filter. Therefore, the audio signals in each frequency band divided by it have a controlled minimum phase characteristic, so that the variable resolution impulse response filter exhibits minimal ringing and excellent transient response in the time domain, thereby making the speaker equalization effect better.

As shown in FIG10 , the following is a waveform effect diagram of the target audio signal obtained by the speaker after filtering with a correlation filter other than the variable resolution impulse response filter (such as a FIR filter (Finite Impulse Response, finite unit impulse response filter) and filtering with a variable resolution impulse response filter. It can be seen that compared with the correlation filter, the variable resolution impulse response filter has much better correction efficiency in high frequency and low frequency correction effect than the traditional filter, and the amount of calculation is 90% smaller than that of the traditional filter with the same effect. In addition, the processing of high-frequency signals in the audio signal only needs to be correlated with the sampling frequency rather than exponentially increased, so it is particularly suitable for acoustic correction of Hi-Res audio equipment.

Based on this, the phase of the variable resolution impulse response filter is kept at the same phase as the phase of the speaker, so that the audio signal processed by the variable resolution impulse response filter is input into the speaker and processed by the speaker, and the output target audio signal is more balanced, thereby ensuring that the sound quality of the audio output by the audio equipment is more balanced.

In some specific embodiments, as shown in FIG11 , the above-mentioned audio signal processing method is usually implemented in the form of a sound effect optimization function program algorithm and stored in an .ini file format. Specifically, the user inputs a play instruction of the audio file in the interactive interface of the human-computer interaction device to start the optimization processing function of the audio signal. The audio device calls the .ini file to call out the algorithm parameters (such as the second sampling frequency, the frequency of the first harmonic information and the frequency of the second harmonic information, the filter coefficient, etc.), and can also configure the parameter values of the algorithm parameters in the .ini file, and then calls the algorithm parameters and the corresponding parameter values and starts the sound effect optimization function, and finally outputs the optimized audio signal.

In some other specific embodiments, as shown in FIG. 12 , the processing process of the audio signal of the audio device is further explained.

(1) An audio device receives a first audio signal of an audio file.

(2) The audio device determines whether the user turns on the sound optimization processing function.

(3) If yes, then enter the sound effect optimization processing process (4), if not, output the first audio signal.

(4) The audio device determines whether the first audio signal includes an audio signal in another format. If yes, the process proceeds to (5); if no, the process proceeds to (6).

(5) If yes, the audio device converts the audio signal in other formats into a first audio signal in a dual-channel pulse code modulation format.

(6) If not, the audio device converts the first audio signal into a second audio signal through a sampling rate converter.

(7) The audio device performs medium and high frequency harmonic compensation on the second audio signal through the first harmonic generator to generate a third audio signal.

(8) The audio device performs mid- and high-frequency harmonic compensation on the third audio signal through a second harmonic generator to generate a fourth audio signal.

(9) The audio device generates a target audio signal based on the fourth audio signal.

It can be seen that the above mainly introduces the solution provided by the embodiment of the present application from the perspective of the method. In order to achieve the above functions, the embodiment of the present application provides a hardware structure and/or software module corresponding to each function. It should be easily appreciated by those skilled in the art that, in combination with the modules and algorithm steps of each example described in the embodiment disclosed herein, the embodiment of the present application can be implemented in the form of hardware or a combination of hardware and computer software. Whether a function is executed in the form of hardware or computer software driving hardware depends on the specific application and design constraints of the technical solution. Professional and technical personnel can use different methods to implement the described functions for each specific application, but such implementation should not be considered to exceed the scope of the present invention.

The embodiment of the present application can divide the controller into functional modules according to the above method example. For example, each functional module can be divided according to each function, or two or more functions can be integrated into one processing module. The above integrated module can be implemented in the form of hardware or in the form of software functional modules. Optionally, the division of modules in the embodiment of the present application is schematic and is only a logical function division. There may be other division methods in actual implementation.

The embodiment of the present application also provides a schematic diagram of the hardware structure of a controller. As shown in Figure 13, the controller 300 includes a processor 301, and optionally, also includes a memory 302 and a communication interface 303 connected to the processor 301. The processor 301, the memory 302 and the communication interface 303 are connected via a bus 304.

The processor 301 may be a central processing unit (CPU), a general-purpose processor network processor (NP), a digital signal processor (DSP), a microprocessor, a microcontroller, a programmable logic device (PLD), or any combination thereof. The processor 301 may also be any other device having a processing function, such as a circuit, a device, or a software module. The processor 301 may also include multiple CPUs, and the processor 301 may be a single-core (single-CPU) processor or a multi-core (multi-CPU) processor. The processor here may refer to one or more devices, circuits, or processing cores for processing data (such as computer program instructions).

The memory 302 may be a read-only memory (ROM) or other types of static storage devices that can store static information and instructions, a random access memory (RAM) or other types of dynamic storage devices that can store information and instructions, or an electrically erasable programmable read-only memory (EEPROM), a compact disc read-only memory (CD-ROM) or other optical disc storage, optical disc storage (including compressed optical disc, laser disc, optical disc, digital versatile disc, Blu-ray disc, etc.), a disk storage medium or other magnetic storage device, or any other medium that can be used to carry or store the desired program code in the form of an instruction or data structure and can be accessed by a computer, and the present embodiment of the application does not impose any restrictions on this. The memory 302 may exist independently or be integrated with the processor 301. Among them, the memory 302 may contain a computer program code. The processor 301 is used to execute the computer program code stored in the memory 302, so as to realize the control method provided in the embodiment of the present application.

The communication interface 303 can be used to communicate with other devices or communication networks (such as Ethernet, radio access network (RAN), wireless local area networks (WLAN), etc. The communication interface 303 can be a module, a circuit, a transceiver or any device that can achieve communication.

The bus 304 may be a peripheral component interconnect (PCI) bus or an extended industry standard architecture (EISA) bus, etc. The bus 304 may be divided into an address bus, a data bus, a control bus, etc. For ease of representation, FIG13 only uses one thick line, but does not mean that there is only one bus or one type of bus.

The embodiment of the present application further provides a computer-readable storage medium, including computer-executable instructions, which, when executed on a computer, enable the computer to execute any one of the audio signal processing methods for an audio device provided in the above embodiments.

The embodiment of the present application further provides a computer program product including computer-executable instructions, which, when executed on a computer, enables the computer to execute any one of the audio signal processing methods for an audio device provided in the above embodiments.

In the above embodiments, it can be implemented in whole or in part by software, hardware, firmware or any combination thereof. When implemented using a software program, it can be implemented in whole or in part in the form of a computer program product. The computer program product includes one or more computer-executable instructions. When the computer-executable instructions are loaded and executed on a computer, the process or function according to the embodiment of the present application is generated in whole or in part. The computer can be a general-purpose computer, a special-purpose computer, a computer network, or other programmable device. The computer-executable instructions can be stored in a computer-readable storage medium, or transmitted from one computer-readable storage medium to another computer-readable storage medium. For example, the computer-executable instructions can be transmitted from a website site, computer, server or data center by wired (e.g., coaxial cable, optical fiber, digital subscriber line (digital subscriber line, DSL)) or wireless (e.g., infrared, wireless, microwave, etc.) mode to another website site, computer, server or data center. The computer-readable storage medium can be any available medium that can be accessed by a computer or a data storage device such as a server, data center, etc. that contains one or more servers that can be integrated with the medium. The available medium may be a magnetic medium (eg, a floppy disk, a hard disk, a magnetic tape), an optical medium (eg, a DVD), or a semiconductor medium (eg, a solid state disk (SSD)).

Although the present application is described herein in conjunction with various embodiments, in the process of implementing the claimed application, those skilled in the art may understand and implement other variations of the disclosed embodiments by viewing the drawings, the disclosure, and the appended claims. In the claims, the word "comprising" does not exclude other components or steps, and "one" or "an" does not exclude multiple situations. A single processor or other unit may implement several functions listed in a claim. Certain measures are recorded in different dependent claims, but this does not mean that these measures cannot be combined to produce good results.

Although the present application has been described in conjunction with specific features and embodiments thereof, it is obvious that various modifications and combinations may be made thereto without departing from the spirit and scope of the present application. Accordingly, this specification and the drawings are merely exemplary illustrations of the present application as defined by the appended claims, and are deemed to have covered any and all modifications, variations, combinations or equivalents within the scope of the present application. Obviously, those skilled in the art may make various modifications and variations to the present application without departing from the spirit and scope of the present application. Thus, if these modifications and variations of the present application fall within the scope of the claims of the present application and their equivalents, the present application is also intended to include these modifications and variations.

The above is only a specific implementation of the present application, but the protection scope of the present application is not limited thereto. Any changes or substitutions within the technical scope disclosed in the present application should be included in the protection scope of the present application. Therefore, the protection scope of the present application should be based on the protection scope of the claims.

Claims (10)

1. An audio device, characterized in that it comprises a sampling rate converter, a primary harmonic generator, a secondary harmonic generator, a speaker and a controller; the controller is connected to the sampling rate converter, the primary harmonic generator, the secondary harmonic generator and the speaker; The controller is configured to: In response to a user's play operation on an audio file, controlling the sampling rate converter to convert a first audio signal of a first sampling frequency in the audio file into a second audio signal of a second sampling frequency, wherein the first sampling frequency is less than the second sampling frequency, and the sampling amplitude of the first audio signal is less than the sampling amplitude of the second audio signal; Controlling the primary harmonic generator to generate first harmonic information according to the first audio signal, and performing harmonic compensation on the second audio signal according to the first harmonic information to generate a third audio signal; controlling the secondary harmonic generator to generate second harmonic information according to the third audio signal, and performing harmonic compensation on the third audio signal according to the second harmonic information to generate a fourth audio signal; The speaker is controlled to output a target audio signal according to the fourth audio signal, wherein a frequency of the first harmonic information is lower than a frequency of the second harmonic information. 2. The audio device according to claim 1, wherein the first audio signal is an audio signal in a dual-channel pulse code modulation format, and before controlling the sampling rate converter to convert the first audio signal of the first sampling frequency in the audio file into a second audio signal of the second sampling frequency, the audio device further comprises: In the case that the audio file includes audio signals in other formats, the audio signals in other formats are converted into audio signals in a two-channel pulse code modulation format, wherein the other formats include formats other than the two-channel pulse code modulation format. 3. The audio device according to claim 1, wherein the performing harmonic compensation on the second audio signal according to the first harmonic information to generate the third audio signal comprises: According to the first harmonic information, harmonic compensation is performed on an audio signal within a first frequency band in the second audio signal to obtain the third audio signal, wherein the frequency of the first harmonic information belongs to the first frequency band, and the first frequency band is a frequency range in which a sampling frequency is less than the first sampling frequency. 4. The audio device according to claim 1, wherein the third audio signal is subjected to harmonic compensation according to the second harmonic information to generate a fourth audio signal, comprising: converting the frequencies of the audio signals in the second frequency band of the third audio signal into frequencies greater than or equal to the second sampling frequency; the second frequency band is a frequency range between the first sampling frequency and the second sampling frequency, and the frequency of the second harmonic information belongs to the second frequency band; According to the second harmonic information, harmonic compensation is performed on the audio signal in the third audio signal whose frequency is greater than or equal to the second sampling frequency to obtain the fourth audio signal. 5. The audio device according to claim 1, characterized in that the audio device further comprises: a variable resolution impulse response filter; the variable resolution impulse response filter is connected to the controller; Before controlling the speaker to output the target audio signal according to the fourth audio signal, the controller is further configured to: Controlling the variable resolution impulse response filter to divide the fourth audio signal into a plurality of audio signals in different frequency bands; According to the filtering parameters corresponding to the respective frequency bands, the audio signals of the respective frequency bands are respectively filtered to obtain a fifth audio signal, wherein the fifth audio signal includes an audio signal obtained by filtering the audio signals of the plurality of frequency bands; Generate the target audio signal of a preset frequency band according to the fifth audio signal; the preset frequency band is a frequency range lower than the first sampling frequency; The controller controls the speaker to output a target audio signal according to the fourth audio signal, and is configured to: The speaker is controlled to output the target audio signal in the preset frequency band. 6 . The audio device according to claim 5 , wherein a phase of the speaker is the same as a phase of the variable resolution impulse response filter. 7. The audio device according to any one of claims 1 to 6, characterized in that before controlling the sampling rate converter to convert the first audio signal of the first sampling frequency in the audio file into the second audio signal of the second sampling frequency, it further comprises: The audio file is parsed to obtain the first audio signal. 8. A method for processing an audio signal, characterized in that the method comprises: In response to a user's play operation on an audio file, converting a first audio signal of a first sampling frequency in the audio file into a second audio signal of a second sampling frequency, wherein the first sampling frequency is lower than the second sampling frequency, and a sampling amplitude of the first audio signal is lower than a sampling amplitude of the second audio signal; generating first harmonic information according to the first audio signal, and performing harmonic compensation on the second audio signal according to the first harmonic information to generate a third audio signal; generating second harmonic information according to the third audio signal, and performing harmonic compensation on the third audio signal according to the second harmonic information to generate a fourth audio signal; The speaker is controlled to output a target audio signal according to the fourth audio signal, wherein a frequency of the first harmonic information is smaller than a frequency of the second harmonic information. 9. The audio signal processing method according to claim 8, wherein the performing harmonic compensation on the second audio signal according to the first harmonic information to generate a third audio signal comprises: According to the first harmonic information, harmonic compensation is performed on an audio signal within a first frequency band in the second audio signal to obtain the third audio signal, wherein the frequency of the first harmonic information belongs to the first frequency band, and the first frequency band is a frequency range in which a sampling frequency is less than the first sampling frequency. 10. The audio signal processing method according to claim 8, characterized in that performing harmonic compensation on the third audio signal according to the second harmonic information to generate a fourth audio signal comprises: converting the sampling frequencies of the audio signals in the second frequency band of the third audio signal into the second sampling frequency; the second frequency band is a frequency range between the first sampling frequency and the second sampling frequency, and the frequency of the second harmonic information belongs to the second frequency band; According to the second harmonic information, harmonic compensation is performed on the audio signal of the second sampling frequency in the third audio signal to obtain the fourth audio signal.