CN104503297A - DSP audio digital signal processing system and method - Google Patents

Abstract

The present invention relates to the field of electroacoustic technology, and in particular to a DSP audio digital signal processing system and method. The DSP audio digital signal processing system includes an audio analog signal input interface module, a DSP digital signal processing module, an audio analog signal output interface module, Single-chip micro-control module, display module and user operation interface module, the display module, user-operation interface module and DSP digital signal processing module are all connected with single-chip micro-control module signal, the audio analog signal input interface module and audio analog signal output The interface modules are all connected to the DSP digital signal processing module signal. With this system and method, the traditional pure analog signal processing audio power amplifier pre-stage can be changed into a digital pre-amplifier with DSP digital signal processing function. , efficiency and flexibility.

Description

DSP audio digital signal processing system and method

technical field

The invention relates to the field of electroacoustic technology, in particular to a DSP audio digital signal processing system and method.

Background technique

Audio electrical signals are converted into actual sound signals through loudspeakers. As a transducer device, its performance has the greatest impact on sound quality. In audio equipment, the speaker is the most important device and also the weakest device. Although there are many types, the structural principles are basically the same. In addition, due to the different materials, the performance of different speakers varies greatly. Inherent defects in technology cannot be completely remedied by updating the design or material selection. The speaker is the terminal equipment of the entire audio system, and its sound-generating component is the speaker. Therefore, the performance of the speaker determines the performance of the speaker to a large extent. Speakers can be divided into active speakers (also called active speakers) and passive speakers (also called passive speakers). Active speakers have a dedicated power amplifier circuit inside, which is directly driven by a lower audio signal level. Passive speakers require an external power amplifier. The present invention is aimed at the active sound box, therefore, the loud sound box in all documents all refers to the active sound box. The traditional power amplifier uses a pure analog circuit. In addition to amplifying the audio signal, if you want to correct the frequency response performance of the speaker, you can only achieve it by adding hardware circuits. Due to the influence of circuit principles and component parameters, it is difficult to achieve ideal effect, and only some simple adjustments can be made. If you want to achieve precise adjustment, the complexity of the circuit will be greatly increased.

Contents of the invention

The technical problem to be solved by the present invention is to provide a DSP audio digital signal processing system and method, which can change the traditional pure analog signal processing audio power amplifier pre-stage into a digital audio amplifier with DSP digital signal processing function. Relying on the high speed, high efficiency and flexibility of digital signal processing, the front stage can make up for the inherent defects of speakers, speakers and audio power amplifiers from the software level, and greatly improve the performance parameters and auditory effects of the audio system.

The technical solution adopted in the present invention is: a DSP audio digital signal processing system and method,

The DSP audio digital signal processing system includes an audio analog signal input interface module, a DSP digital signal processing module, an audio analog signal output interface module, a single-chip micro-control module, a display module and a user operation interface module, the display module, the user operation interface The module and the DSP digital signal processing module are all signal-connected with the microcontroller micro-control module, and the audio analog signal input interface module and the audio analog signal output interface module are all signal-connected with the DSP digital signal processing module;

The audio analog signal input interface module is used to connect external audio signal input, signal amplification and improve signal-to-noise ratio, and the processed signal is sent to the analog signal input interface of the DSP digital signal processing module;

The DSP digital signal processing module is used to perform A/D conversion and digital signal processing on the input analog audio signal, and then output the frequency-divided treble, bass and subwoofer analog signals to the audio through D/A conversion. Analog signal output interface module;

The audio analog signal output interface module is divided into treble output channel, bass output channel and subwoofer output channel. Directly used as the signal source of the subwoofer;

The display module is used as a user interface interface for displaying system functions, module operating status, parameters and user operation instructions;

The user operation interface module integrates all operations and settings of the system;

The single-chip micro-control module is used to control the operating status of other modules in the system; the single-chip micro-control program in the single-chip micro-control module is integrated with program codes and parameters of DSP digital signal processing, and the main functions include DSP digital signal processing program Loading of codes and parameters, operation status control and operation parameter setting update of each functional module of DSP digital signal processing, LCD display control and update, user operation detection and identification, system operation status and system power-off monitoring;

The DSP digital signal processing module includes an A/D conversion module, a loudspeaker frequency response curve correction module and a D/A conversion module connected in sequence;

The DSP audio digital signal processing method mainly comprises the following steps:

(1) Set the speaker frequency response curve correction parameters and set other parameters through the user operation interface module;

(2) The external audio analog signal is input to the audio analog signal input interface module, and then input to the A/D conversion module after the signal is amplified by the audio analog signal input interface module and the signal-to-noise ratio is improved, and then converted into The audio digital signal is input to the speaker frequency response curve correction module;

(3) The speaker frequency response curve correction module processes the transmitted audio digital signal according to the speaker frequency response curve correction parameters set in step (1);

(4), input the audio digital signal processed in step (3) to the D/A conversion module, and then convert it into an audio analog signal through the D/A conversion module;

(5) Outputting the audio analog signal converted in step (4) through the audio analog signal output interface module;

And the step (1) of setting speaker frequency response curve correction parameters mainly includes the following steps:

A. First use electro-acoustic testing equipment to measure the actual frequency response curve and actual phase response curve of the speaker in a standard anechoic chamber;

B. Compare the actual frequency response curve measured in step A with the target frequency response curve formulated by the speaker engineer and speaker engineer to obtain the difference at each frequency point of the audio, thus obtaining a speaker frequency response curve Correction table, the correction table includes the number of frequency correction points to be inserted, and the center frequency of the filter required for each frequency point, filter type, filter Q value and the amplitude parameters of boost or attenuation;

C. Set the speaker frequency response curve correction module according to the parameters in the speaker frequency response curve correction table obtained in step B;

D. Download the speaker frequency response curve correction module set in step C to the dedicated speaker frequency response curve correction debugging board, and repeat step A to test the corrected speaker to obtain the actual measured value after correction The frequency response curve of the loudspeaker, and compare it with the target frequency response curve established in step B. If there is still a difference between the two, fine-tune the correction table for the frequency response curve of the speaker, and then use the correction table for the frequency response curve of the speaker after fine-tuning Set the parameters in the speaker frequency response curve correction module, and then repeat step D until the two are the same;

E. Adjust the parameters in the speaker frequency response curve correction table completed in step D to set the speaker frequency response curve correction module, and then add the set speaker frequency response curve correction module to the DSP digital signal processing module.

Compared with the prior art by adopting the above structure and method, the present invention has the following advantages: the present invention introduces DSP digital signal processing technology into the audio signal amplification system, and designs a set of audio power amplifier pre-stage system with DSP digital signal processing function, Relying on the powerful digital signal processing capability and programmability of DSP, the performance of the speaker is improved at the software level. This system can completely omit the audio signal processing analog electronic circuit part, and use a DSP digital signal processing chip and simple peripheral circuits. Instead, all functions related to audio signal processing are completed by software, and finally integrated into the DSP chip for implementation. The increase in functions does not require any changes to the hardware, and more fine adjustment and processing can be achieved. In the past, the functions that could not be realized by relying on pure hardware circuits can be easily realized in this system. At the same time, a single-chip micro-control system and display function are added to the system, and all operations are integrated into a rotary encoder with buttons, which makes the whole system more intelligent and displayable, and greatly improves the man-machine operability. For the whole speaker, the performance is closer to the idealized requirement.

Description of drawings

Fig. 1 is a system block diagram of the DSP audio digital signal processing system and method of the present invention.

Fig. 2 is a diagram of the start-up steps of the DSP audio digital signal processing system and method of the present invention.

Fig. 3 is a flow chart of the system setting operation of the DSP audio digital signal processing system and method of the present invention.

Fig. 4 is a comparison result of the frequency response curve of the sound box of the DSP audio digital signal processing system of the present invention before and after correction.

Fig. 5 is the optional sound effect of the speakers of the DSP audio digital signal processing system of the present invention and their corresponding frequency response curves.

Fig. 6 is a comparison diagram of the processing effect performed by the speakers of the DSP audio digital signal processing system of the present invention when the installation positions are different.

FIG. 7 is a processing effect diagram of the loudspeaker and bass processing modules of the DSP audio digital signal processing system of the present invention.

Fig. 8 shows the effect of the subwoofer speaker on frequency response curve correction and the setting of the subwoofer cut-off frequency of the DSP audio digital signal processing system of the present invention.

Detailed ways

The present invention will be further described below in conjunction with the accompanying drawings and specific embodiments, but the present invention is not limited to the following specific embodiments.

As shown in the figure: a DSP audio digital signal processing system and method,

The DSP audio digital processing system includes an audio analog signal input interface module, a DSP digital signal processing module, an audio analog signal output interface module, a single-chip micro-control module, a display module and a user operation interface module, the display module, the user operation interface module And the DSP digital signal processing module is all connected with the single-chip micro-control module signal, and the audio analog signal input interface module and the audio analog signal output interface module are all connected with the DSP digital signal processing module signal;

The audio analog signal input interface module is used to connect external audio signal input, signal amplification and improve signal-to-noise ratio, and the processed signal is sent to the analog signal input interface of the DSP digital signal processing module;

The DSP digital signal processing module is used to perform A/D conversion and digital signal processing on the input analog audio signal, and then output the frequency-divided treble, bass and subwoofer analog signals to the audio through D/A conversion. Analog signal output interface module;

The audio analog signal output interface module is divided into treble output channel, bass output channel and subwoofer output channel. Directly used as the signal source of the subwoofer;

The display module is used as a user interface interface for displaying system functions, module operating status, parameters and user operation instructions; the display module is an LCD display;

The user operation interface module integrates all operations and settings of the system; the user operation interface module is a rotary encoder with button function;

The single-chip micro-control module is used to control the operating status of other modules in the system;

The DSP digital signal processing module includes an A/D conversion module, a loudspeaker frequency response curve correction module and a D/A conversion module connected in sequence; the DSP digital signal processing module also includes an input signal pressure limit control module, a system master volume Control and system mute control module, noise processing module, audio treble processing module, audio bass processing module, loudspeaker loudness control module, sound effect setting module, system delay module, speaker maximum power control module, audio treble and bass frequency division module and subwoofer Set up modules.

The DSP audio digital signal processing method mainly comprises the following steps:

(1) Set the speaker frequency response curve correction parameters and set other parameters through the user operation interface module;

(2) The external audio analog signal is input to the audio analog signal input interface module, and then input to the A/D conversion module after the signal is amplified by the audio analog signal input interface module and the signal-to-noise ratio is improved, and then converted into The audio digital signal is input to the speaker frequency response curve correction module;

(3) The speaker frequency response curve correction module processes the transmitted audio digital signal according to the speaker frequency response curve correction parameters set in step (1);

(4), input the audio digital signal processed in step (3) to the D/A conversion module, and then convert it into an audio analog signal through the D/A conversion module;

(5) Outputting the audio analog signal converted in step (4) through the audio analog signal output interface module.

And before step (3), it is necessary to limit the maximum peak value of the input audio signal: when the maximum peak value of the input audio signal exceeds the threshold, the audio signal is compressed according to the signal peak compression table, and is displayed on the LCD according to the degree of compression Display on the screen to prompt the user to reduce the volume of the input signal; for signals that do not exceed the threshold, no processing is performed.

Then perform system master volume control and system mute control: divide the system volume into 91 levels: -80dB ~ +10dB, and adjust the step size by 1dB, set by the user, when the volume is less than -79dB, the system will automatically switch to mute model.

After step (3), audio treble processing is also required: use a high shoulder filter or a peak filter to boost or attenuate the treble, the adjustment range is: -10dB ~ +10dB, and the adjustment is made in steps of 1dB.

Audio bass processing: use low shoulder filter or peak filter to boost or attenuate the bass, the adjustment range is: -10dB ~ +10dB, and adjust with 1dB as the step size.

Loudspeaker loudness control: When the output sound pressure level of the speaker is low, in order to balance the loudness of the high, middle and bass, according to the standard equal loudness curve, the bass part less than 60Hz and the treble part greater than 7000Hz are boosted.

Sound effect setting: Use 10-band equalization filter to boost or attenuate each frequency band of the sound to simulate the auditory effect of the speaker in different applications.

System delay: In a multi-speaker application environment, in order to make the sound output by multiple speakers reach a certain hearing area synchronously, according to the propagation speed of the sound in the air, the speakers in different positions are delayed according to the distance before outputting , and the delay time is set by the user in the form of distance (unit: m (meter) or ft (feet)).

Speaker maximum power control: According to the maximum output power requirements of the speaker, limit the instantaneous peak level of the signal output from the audio analog signal output interface to the rear stage of the audio power amplifier to ensure that the power amplifier works within a safe power range and avoid burning out due to excessive power The power amplifier or speaker compresses the level exceeding the specified threshold proportionally, and the compressed signal level = original signal level * (threshold level / original quotation mark level). Signals that do not exceed the threshold are not processed.

Audio high and low frequency division: In addition to high and low two-frequency output, high, medium and low three-frequency divisor can also be performed, the filter type, crossover point frequency, end gain, output polarity of the frequency divider of each frequency band Parameters such as are optional, and the specific requirements are determined by the engineer, but they are invisible to the user.

Subwoofer setting: The process is similar to the audio treble and bass crossover. The filter type, cutoff frequency, end gain, output polarity and other parameters of the subwoofer crossover are optional. The specific requirements are determined by the engineer, but for the user, it is not necessary visible.

And the step (1) of setting speaker frequency response curve correction parameters mainly includes the following steps:

A. First use electro-acoustic testing equipment to measure the actual frequency response curve and actual phase response curve of the speaker in a standard anechoic chamber;

B. Compare the actual frequency response curve measured in step A with the target frequency response curve formulated by the speaker engineer and speaker engineer to obtain the difference at each frequency point of the audio, thus obtaining a speaker frequency response curve Correction table, the correction table includes the number of frequency correction points to be inserted, and the center frequency of the filter required for each frequency point, filter type, filter Q value and the amplitude parameters of boost or attenuation;

C. Set the speaker frequency response curve correction module according to the parameters in the speaker frequency response curve correction table obtained in step B;

D. Download the speaker frequency response curve correction module set in step C to the dedicated speaker frequency response curve correction debugging board, and repeat step A to test the corrected speaker to obtain the actual measured value after correction The frequency response curve of the loudspeaker, and compare it with the target frequency response curve established in step B. If there is still a difference between the two, fine-tune the correction table for the frequency response curve of the speaker, and then use the correction table for the frequency response curve of the speaker after fine-tuning Set the parameters in the speaker frequency response curve correction module, and then repeat step D until the two are the same;

E. Adjust the parameters in the speaker frequency response curve correction table completed in step D to set the speaker frequency response curve correction module, and then add the set speaker frequency response curve correction module to the DSP digital signal processing module.

As shown in Fig. 1, it is a block diagram of the system structure of the present invention. Among them, the LCD display and the rotary encoder with buttons are the user interface. All the user's operation settings on the system are realized by rotating the rotary encoder clockwise, counterclockwise or pressing the buttons. The LCD display is used to display System functions and the operating status and parameters of each functional module. At the same time, when the user sets the system, the user will be prompted for the operation; the audio analog signal input interface receives the external audio signal input and performs pre-noise reduction and amplification processing, and then sends the signal to the To the analog input port of the DSP digital signal processing module; the DSP digital signal processing module has its own analog/digital conversion and digital/analog conversion, the analog/digital conversion module converts the audio analog signal into a digital signal, and the digital/analog conversion module will be processed The final digital audio signal is re-converted into an analog signal, and then sent to the audio power amplifier for further amplification. All operations related to audio processing in this system are completed in the DSP digital signal processing module; the microcontroller module is the control of the entire system. Part, after the system is powered on, firstly control the reset of each module in the system, then load the DSP program code and parameters into the DSP digital signal processor through the I2C bus, and at the same time control the LCD display through the SPI bus to complete the system startup according to the process The display step is to control the LCD to display the main interface of the system after the system enters the normal working state, and then start to accept user operation settings.

Figure 2 shows the startup steps of the system: first, power on the system; after the system is powered on, the microcontroller module first resets, and then controls other modules in the system to reset after the reset is completed; wait for all modules in the system to complete the reset Control the LCD to display the company Logo, and then load the DSP digital signal processing program code and parameter data into the DSP digital signal processing chip, but at this time the DSP is in a mute state, wait for 2S and then control the LCD to display the product model, and then wait for another 2S Control the LCD to display the main interface of the system, and at the same time control the DSP digital signal processing module to start working, the whole system starts to enter the normal working state, and the microcontroller begins to accept user operations; in the state of the main interface of the system, the user can turn left or right Rotate the encoder to adjust the system master volume.

Figure 3 is the flow chart of the system setting: In the state of the main interface of the system, press the button of the rotary encoder to enter the system parameter setting state, and all the functions in the system that can be set by the user are listed in Figure 3 modules and their corresponding optional parameter values, and for the frequency response curve correction module, loudness control module, signal pressure limit processing module, power amplifier maximum power monitoring module, high-bass crossover module, etc., it is completely designed to improve the quality of the speaker Yes, for different speakers and speakers, the specific functions and parameters of the module are the optimal solution for fixed selection, which is invisible to the user, so the user cannot set these parameters.

After entering the system parameter setting state, select different function menus by rotating the rotary encoder left or right. Rotating the rotary encoder to the right by one gear will select the next menu item, while rotating it to the left by one gear will select the previous menu item, and the selected menu item will be displayed in reverse, and the corresponding parameter on the right side of the menu item The value is displayed normally. When it is necessary to modify the parameter value of a function item, firstly rotate the rotary encoder to select the menu label of the function item, and then press the rotary encoder to enter the parameter setting state of the menu item. At this time, the menu item The label of the menu item returns to the normal display state, and the current parameter value of the menu item on the right is flipped and displayed, and the parameter value is displayed on the left to be displayed in the center, and the left and right inverted triangle guide signs are added. When the current parameter value is on the left side When there are optional parameters, display the inverted triangle guide sign on the left, otherwise it will not display; when there are optional parameters on the right side of the current parameter value, display the inverted triangle guide sign on the right, otherwise it will not display. The optional parameter value list of the menu item is displayed sequentially from left to right or right to left for the user to choose.

When changing the parameter value of a menu item, it takes effect immediately, that is to say, every adjustment made to the parameter value of a menu item will immediately correspond to a state switch of the function module corresponding to the menu item in the system, but It is only a temporary working state. Only when exiting the system parameter setting state, the system will save all the modifications made, and it will be the default working state of the module when the module is turned on next time. After setting the parameters of a menu item, you need to press the rotary encoder to exit the parameter value setting state of the menu item, and the system will automatically return to the selected state of the menu item. At this time, you can continue to modify the parameter values of other function menus. The method of adjusting parameter values of all menu items is the same.

When performing a system reset, that is to restore the system to the working state of each functional module of the system at the factory, first operate the rotary encoder to select the system reset menu, and then press the rotary encoder, the system will control the LCD to display the system reset confirmation dialog interface for The user confirms, at this time, the "YES" or "NO" label can be selected by rotating the rotary encoder left or right, and then press the rotary encoder to notify the system to perform a system reset operation, if the selection is "NO" , the system returns directly to the selected state of the system reset menu without any processing, and the reset operation ends; if the selection is "YES", the system starts to execute the reset operation, and restores the state of all functional modules in the system to the factory state , and then automatically return to the selected state of the system reset menu, and the reset operation ends.

Exit the system parameter setting state. After completing the system parameter setting, you need to exit the system parameter setting state and return to the main interface of the system, you need to first operate the rotary encoder to select the exit and save menu item, and then press the rotary encoder, the system will first change each menu Save the parameter value of the item, and automatically return to the display state of the main interface of the system after saving.

The meanings of the labels in the system setting operation flowchart shown in Figure 3 correspond to the following:

Press : Indicates that the rotary encoder is clicked

Left : Indicates that the rotary encoder rotates to the left

Right : Indicates that the rotary encoder rotates to the right

Default value: Indicates the default parameter value used by the menu item when the system leaves the factory or after performing a system reset

Power-on value: Indicates that the parameter value is used as the default setting every time the system is powered on, and the adjustment of the parameter value of this menu item in this normal working state will not affect the working state when the system is powered on next time.

Speaker frequency response curve correction: This module is aimed at specific models of speakers, and its parameters are jointly debugged and given by speaker design engineers, speaker design engineers and sound test engineers. It is a fixed functional module, so it is invisible to users, and users cannot Adjust its parameters.

As shown in Figure 4 above, it is a comparison result of the frequency response curve of a speaker using DSP audio digital signal processing before and after correction. Curve 1 is the frequency response curve of the speaker bass unit before correction. After correction It becomes curve 5; curve 2 is the frequency response curve of the speaker tweeter before correction, and becomes curve 6 after correction. The module uses nine independent spike filters, and the parameters of each filter are shown in Table 1 below:

serial number Center frequency: Hz Quality factor Q value Boost or attenuation range: dB filter type 1 70.00 1.5 6.0 Peaking Filter 2 160.00 3.0 1.0 Peaking Filter 3 550.00 2.0 -5.0 Peaking Filter 4 1200.00 10.0 -3.0 Peaking Filter 5 1700.00 5.0 -2.0 Peaking Filter 6 3700.00 2.0 -4.0 Peaking Filter 7 5500.00 5.0 -3.0 Peaking Filter 8 8700.00 5.0 3.0 Peaking Filter 9 13200.00 5.0 5.0 Peaking Filter

Table 1

In the boost or attenuation range in the table, a positive value means boosting the signal of the frequency band, and a negative value means attenuating the signal of the frequency band. Treble and bass frequency division, the cut-off frequency of bass is 2700.00Hz, using Linkwitz-Riley (Linkwitz-Riley 48) 48-order filter, the end gain is 0dB; the starting frequency of treble is 2600.00Hz, also using Linkwitz-Riley -Riley (Linkwitz-Riley 48) 48th order filter with an end gain of 2.0dB.

Figure 5 shows the optional sound effects and corresponding frequency response curves of a speaker that uses DSP audio digital signal processing pre-stage. The processing method is also to boost or attenuate a certain frequency band of the audio signal, but a 10-segment parametric equalizer is used to construct a virtual auditory environment effect. No. 10 curve MUSIC in the figure does not do any processing to the signal, for The default effect state of the system; curve 11 LIVE builds the auditory effect of the concert scene; curve 12 CLUB builds the auditory effect in the club; line 13 SPEECH is the processing effect of the voice signal when the microphone is connected .

Figure 6 shows a comparison of the processing effects of a speaker using DSP audio digital signal processing pre-stage when the installation location is different. For example, in the POLE bracket installation mode, the signal below 1kHz is boosted, and the signal between 60 and 500Hz is boosted by about 6dB, which is used to compensate for the weakening of the bass caused by the sound spreading to the back of the speaker.

As shown in Figure 7, the processing effect of a speaker high and bass processing module using DSP audio digital signal processing front stage, a peak filter is used for bass processing, the center frequency is 65.00Hz, and the quality factor (Q value) is 1.0, the end gain is 0; a peak filter is also used for treble processing, the center frequency is 11000.00Hz, the quality factor (Q value) is 5.0, and the end gain is 0. The boost or attenuation range of treble and bass is +10dB ~ -10dB, the curve in the middle is the curve without any processing (0dB), the curve after the maximum boost of 10dB, and the curve after the maximum attenuation of -10dB . The treble and bass modules are designed to meet personal listening preferences.

Figure 8 shows the effect of frequency response curve correction and the setting of subwoofer cutoff frequency by a subwoofer speaker using DSP audio digital signal processing pre-stage. Curve No. 6 is the original frequency response curve (without DSP audio digital signal processing), and No. 8 curve is the corrected effect of frequency response curve with DSP audio digital signal processing. The subwoofer cutoff frequency setting uses a Linkwitz-Riley 48 order filter, the cutoff frequency of curve 10 is 150.00Hz, the cutoff frequency of curve 11 is 120.00Hz, and the cutoff frequency of curve 12 is 100.00Hz, the cut-off frequency of curve No. 13 is 80.00Hz, and the cut-off frequency of curve No. 14 is 60.00Hz. The specific parameters are set by the user.

Claims (3)

1. A DSP audio digital signal processing system, characterized in that: The DSP audio digital signal processing system includes an audio analog signal input interface module, a DSP digital signal processing module, an audio analog signal output interface module, a single-chip micro-control module, a display module and a user operation interface module, the display module, the user operation interface The module and the DSP digital signal processing module are all signal-connected with the microcontroller micro-control module, and the audio analog signal input interface module and the audio analog signal output interface module are all signal-connected with the DSP digital signal processing module; The audio analog signal input interface module is used to connect external audio signal input, signal amplification and improve signal-to-noise ratio, and the processed signal is sent to the analog signal input interface of the DSP digital signal processing module; The DSP digital signal processing module is used to perform A/D conversion and digital signal processing on the input analog audio signal, and then output the frequency-divided treble, bass and subwoofer analog signals to the audio through D/A conversion. Analog signal output interface module; The audio analog signal output interface module is divided into treble output channel, bass output channel and subwoofer output channel. Directly used as the signal source of the subwoofer; The display module is used as a user interface interface for displaying system functions, module operating status, parameters and user operation instructions; The user operation interface module integrates all operations and settings of the system; The single-chip micro-control module is used to control the operating status of other modules in the system; The DSP digital signal processing module includes an A/D conversion module, a loudspeaker frequency response curve correction module and a D/A conversion module connected sequentially. 2. A DSP audio digital signal processing method is characterized in that it comprises the following steps: (1) Set the speaker frequency response curve correction parameters and set other parameters through the user operation interface module; (2) The external audio analog signal is input to the audio analog signal input interface module, and then input to the A/D conversion module after the signal is amplified by the audio analog signal input interface module and the signal-to-noise ratio is improved, and then converted into The audio digital signal is input to the speaker frequency response curve correction module; (3) The speaker frequency response curve correction module processes the transmitted audio digital signal according to the speaker frequency response curve correction parameters set in step (1); (4), input the audio digital signal processed in step (3) to the D/A conversion module, and then convert it into an audio analog signal through the D/A conversion module; (5) Outputting the audio analog signal converted in step (4) through the audio analog signal output interface module. 3. A kind of DSP audio digital signal processing method according to claim 2, is characterized in that: the setting speaker frequency response curve correction parameter of described step (1) mainly comprises the following steps: A. First use electro-acoustic testing equipment to measure the actual frequency response curve and actual phase response curve of the speaker in a standard anechoic chamber; B. Compare the actual frequency response curve measured in step A with the target frequency response curve formulated by the speaker engineer and speaker engineer to obtain the difference at each frequency point of the audio, thus obtaining a speaker frequency response curve Correction table, the correction table includes the number of frequency correction points to be inserted, and the center frequency of the filter required for each frequency point, filter type, filter Q value and the amplitude parameters of boost or attenuation; C. Set the speaker frequency response curve correction module according to the parameters in the speaker frequency response curve correction table obtained in step B; D. Download the speaker frequency response curve correction module set in step C to the dedicated speaker frequency response curve correction debugging board, and repeat step A to test the corrected speaker to obtain the actual measured value after correction The frequency response curve of the loudspeaker, and compare it with the target frequency response curve established in step B. If there is still a difference between the two, fine-tune the correction table for the frequency response curve of the speaker, and then use the correction table for the frequency response curve of the speaker after fine-tuning Set the parameters in the speaker frequency response curve correction module, and then repeat step D until the two are the same; E. Adjust the parameters in the speaker frequency response curve correction table completed in step D to set the speaker frequency response curve correction module, and then add the set speaker frequency response curve correction module to the DSP digital signal processing module.