CN104503297B - DSP audio digital signals processing system and method - Google Patents
DSP audio digital signals processing system and method Download PDFInfo
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- CN104503297B CN104503297B CN201410685104.0A CN201410685104A CN104503297B CN 104503297 B CN104503297 B CN 104503297B CN 201410685104 A CN201410685104 A CN 201410685104A CN 104503297 B CN104503297 B CN 104503297B
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- G—PHYSICS
- G05—CONTROLLING; REGULATING
- G05B—CONTROL OR REGULATING SYSTEMS IN GENERAL; FUNCTIONAL ELEMENTS OF SUCH SYSTEMS; MONITORING OR TESTING ARRANGEMENTS FOR SUCH SYSTEMS OR ELEMENTS
- G05B19/00—Programme-control systems
- G05B19/02—Programme-control systems electric
- G05B19/04—Programme control other than numerical control, i.e. in sequence controllers or logic controllers
- G05B19/042—Programme control other than numerical control, i.e. in sequence controllers or logic controllers using digital processors
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- G—PHYSICS
- G05—CONTROLLING; REGULATING
- G05B—CONTROL OR REGULATING SYSTEMS IN GENERAL; FUNCTIONAL ELEMENTS OF SUCH SYSTEMS; MONITORING OR TESTING ARRANGEMENTS FOR SUCH SYSTEMS OR ELEMENTS
- G05B2219/00—Program-control systems
- G05B2219/20—Pc systems
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- G05B2219/25314—Modular structure, modules
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Abstract
The present invention relates to electroacoustic techniques field,More particularly to a kind of DSP audio digital signals processing system and method,The DSP audio digital signals processing system includes audio analog signals input interface module,DSP digital signal processing modules,Audio analog signals output interface module,Single-chip microcomputer micro-control module,Display module and user interface module,The display module,User interface module and DSP digital signal processing modules are connected with single-chip microcomputer micro-control module signal,The audio analog signals input interface module and audio analog signals output interface module are connected with DSP digital signal processing module signals,The audio frequency power amplifier prime that traditional purely analog is processed can be changed to by the digital prime with DSP digital signal processing functions using this System and method for,By the high speed of Digital Signal Processing,Efficient and flexibility.
Description
Technical field
The present invention relates to electroacoustic techniques field, more particularly to a kind of DSP audio digital signals processing system and method.
Background technology
Audio electrical signal is converted to actual voice signal, is completed by loudspeaker, as a kind of energy transducer, its property
Can the good and bad influence maximum to tonequality.In stereo set, loudspeaker is most important device, while being also most weak device
Part, although species is various, but structural principle is substantially identical, additionally, due to the difference of material, causes different loudspeakers
Performance difference is huge, and its inherent shortcoming on the frequency response is to be made up by updating design or material selection completely
's.Audio amplifier is the terminal device of whole sound system, and its sound-generating element is exactly loudspeaker, so, the performance of loudspeaker also just exists
The performance of audio amplifier is largely determined.Audio amplifier can be divided into active audio amplifier(Also it is active audio amplifier)And passive loudspeaker box(Also cry
Passive speaker)Two classes, carry special power amplification circuit inside active audio amplifier, direct using relatively low audio signal level
Drive, passive loudspeaker box needs external power amplifier.The present invention is directed active audio amplifier, therefore, the sounds for improving in all text gears more
Case refers both to active audio amplifier.Traditional power amplifier uses pure analog circuit, in addition to being amplified to audio signal excessively, if it is desired to audio amplifier
Frequency response performance corrected, can only be realized by increasing hardware circuit, influenceed by circuit theory and component parameters, it is difficult to
Preferable effect is reached, and a little simple adjustment can only be done, if it is desired to realized fine adjustment, will greatly increase realization
The complexity of circuit.
The content of the invention
The technical problems to be solved by the invention are:A kind of DSP audio digital signals processing system and method are provided, are used
The audio frequency power amplifier prime that traditional purely analog is processed can be changed to band DSP Digital Signal Processing work(by this System and method for
The digital prime of energy, by the high speed of Digital Signal Processing, efficient and flexibility, makes up loudspeaker, sound from software view
The inherent shortcoming of case and audio frequency power amplifier, performance parameter and auditory effect to sound system are significantly lifted.
The technical solution adopted in the present invention is:A kind of DSP audio digital signals processing system and method,
The DSP audio digital signals processing system is included at audio analog signals input interface module, DSP data signals
Reason module, audio analog signals output interface module, single-chip microcomputer micro-control module, display module and user interface module,
The display module, user interface module and DSP digital signal processing modules with single-chip microcomputer micro-control module signal
Connection, the audio analog signals input interface module and audio analog signals output interface module with DSP data signals
Processing module signal is connected;
The audio analog signals input interface module is used to connect external audio signal input, signal amplification and improves letter
Make an uproar and compare, and the signal after treatment is sent into the analog input interface of DSP digital signal processing modules;
The DSP digital signal processing modules are used to carrying out A/D conversions to the simulated audio signal being input into, at data signal
Reason, treatment changes output by the high pitch after frequency dividing, bass and subwoofer analog signal to audio frequency simulation by D/A again after terminating
Signal output interface module;
The audio analog signals output interface module point high pitch output channel, bass output channel and subwoofer output are logical
Road, high pitch output channel and bass output channel are connected with power amplifier rear class high pitch, bass amplifying circuit respectively, and subwoofer output is logical
Signal source of the super low sound signal of road output directly as super-bass loudspeaker box;
The display module as user interface, for Display System Function, module running status, parameter and use
Family operation instruction;
The all operations of user interface module integrated system are integrated with setting;
The single-chip microcomputer micro-control module is used for the running status of other modules in control system;The single-chip microcomputer microcontroller
The program code and parameter of DSP Digital Signal Processing are integrated with single-chip microcomputer microcontroller program in module, major function includes
The loading of DSP DSP programs code and parameter, the running status control of each functional module of DSP Digital Signal Processing and
Operational factor sets renewal, LCD display controls and renewal, user's operation detection and identification, system running state and system power failure
Monitoring;
The DSP digital signal processing modules include that the A/D modular converters of signal connection, loudspeaker frequency response curve are rectified successively
Positive module and D/A modular converters;
The DSP audio digital signals processing method is mainly included the following steps that:
(1), setting loudspeaker frequency response curve correction parameter and other specification is set by user interface module;
(2), external audio analog signal input audio analog signals input interface module, then by audio analog signals
Input interface module signal amplifies and is input to A/D modular converters after improving signal to noise ratio, then changes by A/D modular converters again
Into digital audio signal input to loudspeaker frequency response curve rectification module;
(3), loudspeaker frequency response curve rectification module is according to step(1)The loudspeaker frequency response curve correction parameter of setting comes right
The audio digital signals for transmitting are processed;
(4), by step(3)Audio digital signals after treatment are input to D/A modular converters, then by D/A moduluss of conversion
Block conversion becomes audio analog signals;
(5), by step(4)Audio analog signals after conversion are exported by audio analog signals output interface module;
And the step(1)Setting loudspeaker frequency response curve correction parameter mainly include the following steps that:
A, first by electroacoustical instrument, the actual frequency response curve and reality of audio amplifier are measured in the anechoic room of standard
Border phase response curve;
B, actual frequency response curve measured in step A is formulated with by loudspeaker engineer and audio amplifier engineer
Target frequency response curve contrasted, the difference on audio each Frequency point is obtained, so as to draw a loudspeaker frequently
Rate response curve correction table, the correction table includes the number of the frequency correction point being inserted, and needed for each Frequency point
The centre frequency of wave filter, filter type, the range parameter of filter Q and lifting or decay;
C, the parameter in the speaker frequency response plot correction table that step B is obtained are come to loudspeaker frequency response curve
Rectification module is configured;
D, loudspeaker frequency response curve rectification module good set by step C is individually downloaded to special loudspeaker frequency ring
In answering curve to correct debugging board, and repeat step A, to being tested through the audio amplifier after overcorrection, obtains actual after overcorrection
The audio amplifier frequency response curve for measuring, and contrasted with the target frequency response curve of formulation in step B, if both also have
Difference, then be micro-adjusted to speaker frequency response plot correction table, then bent according to the frequency response of micro-adjustment rear speaker
Parameter in line correction table is configured to loudspeaker frequency response curve rectification module, and step step D is repeated afterwards, until both
It is identical;
Parameter in E, the speaker frequency response plot correction table for completing step D adjustment is come to loudspeaker frequency response curve
Rectification module is configured, then the loudspeaker frequency response curve rectification module that will be set is added to DSP digital signal processing modules
In.
Using above structure and method compared with prior art, the present invention has advantages below:The present invention believes DSP numerals
Number treatment technology is incorporated into audio signal amplifying system, designs a set of audio frequency power amplifier with DSP digital signal processing functions
Backing system, by DSP powerful digital signal processing capability and programmability, enters on software view come the performance to audio amplifier
Row lifting, the system can thoroughly save Audio Signal Processing Analogical Electronics part, use one piece of DSP Digital Signal Processing
Chip and simple peripheral circuit replace, and all functions relevant with Audio Signal Processing are all completed using software, most
After be integrated into dsp chip and realized, increase functionally simultaneously need not make any change, and can accomplish to hardware
Finer regulation and treatment, relies on the impossible function of pure hardware circuit in the past, in the present system can be easily real
It is existing.Meanwhile, single-chip microcomputer micro control system and display function are added in systems, it is integrated all to operate in a rotation with button
In encoder, make whole system more intelligent and can display, the man-machine operating performance of significant increase.Whole audio amplifier is come
Say, performance is more nearly idealization and requires.
Brief description of the drawings
Fig. 1 is the system block diagram of DSP audio digital signals processing system of the present invention and method.
Fig. 2 is DSP audio digital signals processing system of the present invention and method system start block diagram.
Fig. 3 DSP audio digital signals processing systems of the present invention set operational flowchart with method system.
Fig. 4 is that the frequency response curve of DSP audio digital signals processing system audio amplifier of the present invention is right before correction and after correction
Compare result.
Fig. 5 is the optional audio and its corresponding frequency response curve of DSP audio digital signals processing system audio amplifier of the present invention.
The treatment effect that Fig. 6 is carried out by DSP audio digital signals processing system audio amplifier of the present invention when installation site is different
Fruit comparison diagram.
Fig. 7 is the treatment effect figure of the high and low sound processing module of DSP audio digital signals processing system audio amplifier of the present invention.
Fig. 8 is the effect that DSP audio digital signals processing system super-bass loudspeaker box of the present invention is corrected to frequency response curve and surpasses
The setting of bass cut frequency.
Specific embodiment
The present invention is described further with specific embodiment below in conjunction with accompanying drawing, but the present invention be not limited only to it is following
Specific embodiment.
As shown in the figure:A kind of DSP audio digital signals processing system and method,
The DSP audio digital processings system includes audio analog signals input interface module, DSP Digital Signal Processing moulds
Block, audio analog signals output interface module, single-chip microcomputer micro-control module, display module and user interface module, it is described
Display module, user interface module and DSP digital signal processing modules are connected with single-chip microcomputer micro-control module signal,
The audio analog signals input interface module and audio analog signals output interface module with DSP Digital Signal Processing moulds
Block signal is connected;
The audio analog signals input interface module is used to connect external audio signal input, signal amplification and improves letter
Make an uproar and compare, and the signal after treatment is sent into the analog input interface of DSP digital signal processing modules;
The DSP digital signal processing modules are used to carrying out A/D conversions to the simulated audio signal being input into, at data signal
Reason, treatment changes output by the high pitch after frequency dividing, bass and subwoofer analog signal to audio frequency simulation by D/A again after terminating
Signal output interface module;
The audio analog signals output interface module point high pitch output channel, bass output channel and subwoofer output are logical
Road, high pitch output channel and bass output channel are connected with the high and low amplifying circuit of power amplifier rear class respectively, and subwoofer output channel is defeated
Signal source of the super low sound signal for going out directly as super-bass loudspeaker box;
The display module as user interface, for Display System Function, module running status, parameter and use
Family operation instruction;The display module is LCD display;
The all operations of user interface module integrated system are integrated with setting;The user interface module
It is a rotary encoder with keypress function;
The single-chip microcomputer micro-control module is used for the running status of other modules in control system;
The DSP digital signal processing modules include that the A/D modular converters of signal connection, loudspeaker frequency response curve are rectified successively
Positive module and D/A modular converters;The DSP digital signal processing modules also include input signal pressure limit control module, system
Master volume control and the Jing Yin control module of system, noise processed module, audio treble processing module, audio bass processing module,
Loudspeaker loudness control module, audio setup module, system delay module, audio amplifier maximum power control module, audio bass high
Frequency division module and subwoofer setup module.
The DSP audio digital signals processing method is mainly included the following steps that:
(1), setting loudspeaker frequency response curve correction parameter and other specification is set by user interface module;
(2), external audio analog signal input audio analog signals input interface module, then by audio analog signals
Input interface module signal amplifies and is input to A/D modular converters after improving signal to noise ratio, then changes by A/D modular converters again
Into digital audio signal input to loudspeaker frequency response curve rectification module;
(3), loudspeaker frequency response curve rectification module is according to step(1)The loudspeaker frequency response curve correction parameter of setting comes right
The audio digital signals for transmitting are processed;
(4), by step(3)Audio digital signals after treatment are input to D/A modular converters, then by D/A moduluss of conversion
Block conversion becomes audio analog signals;
(5), by step(4)Audio analog signals after conversion are exported by audio analog signals output interface module.
And in step(3)Also need to limit the peak-peak of input audio signal before:Work as input audio signal
Peak-peak exceed threshold values when, audio signal is compressed according to signal peak compaction table, and according to compression degree exist
Shown to point out the user should to reduce input signal volume on LCD display;For the signal not less than threshold values, then do not carry out
Any treatment.
Then the control of system master volume and the Jing Yin control of system are carried out again:System sound volume is divided into 91 grades:- 80dB~+
10dB, and be regulation step-length with 1dB, it is configured by user, when volume is less than -79dB, system automatically switches to Jing Yin
Pattern.
In step(3)Afterwards, in addition it is also necessary to carry out audio treble treatment:Using shoulder wave filter high or pinnacle wave filter to high pitch
Lifted or decayed, adjustable range is:- 10dB~+10dB, and be adjusted as step-length using 1dB.
Audio bass treatment:Bass is lifted or decayed using low shoulder wave filter or pinnacle wave filter, adjustable range
For:- 10dB~+10dB, and be adjusted as step-length using 1dB.
Loudspeaker loudness is controlled:When audio amplifier output sound pressure level is relatively low, to balance the loudness of high, medium and low sound, establishing criteria
Equal loudness contour figure, the bass part less than 60Hz and the treble portion more than 7000Hz are lifted.
Audio is set:Using 10 sections of equalization filters, each frequency band of sound is lifted or is decayed to simulate audio amplifier
Auditory effect under different application occasion.
System delay:Under many audio amplifier applied environments, for the synchronous sound for making multiple audio amplifier outputs reaches a certain auditory sensation area
Domain, according to the aerial spread speed of sound, exports again after entering line delay by distance to the audio amplifier of different putting positions, and will
Delay duration is with distance(Unit is m (rice) or ft (foot))Form be configured for user.
Audio amplifier maximum power control:According to the requirement of audio amplifier peak power output, audio analog signals output interface is exported
Signal transient peak level to audio frequency power amplifier rear class is limited, it is ensured that power amplifier is operated in the range of firm power, it is to avoid because
Power is excessive and burns out power amplifier or loudspeaker, is compressed compression in proportion to the level more than prescribed threshold, the letter after compression
Number level=original signal level *(Threshold level/original quotation marks level).Signal not less than threshold values is left intact.
Audio height cent is frequently:Except excessively high and low two frequency dividings output, high, medium and low three frequency division divisor, each frequency range can be also carried out
The parameters such as filter type, frequency dividing dot frequency, end gain, the output polarity of frequency divider are optional, and specific requirement is by engineer
It is determined that, but be sightless for a user.
Subwoofer is set:The similar audio height cent of process frequently, the filter type of subwoofer frequency divider, cut-off frequency,
The parameters such as end gain, output polarity are optional, and specific requirement is determined by engineer, but are sightless for a user.
And the step(1)Setting loudspeaker frequency response curve correction parameter mainly include the following steps that:
A, first by electroacoustical instrument, the actual frequency response curve and reality of audio amplifier are measured in the anechoic room of standard
Border phase response curve;
B, actual frequency response curve measured in step A is formulated with by loudspeaker engineer and audio amplifier engineer
Target frequency response curve contrasted, the difference on audio each Frequency point is obtained, so as to draw a loudspeaker frequently
Rate response curve correction table, the correction table includes the number of the frequency correction point being inserted, and needed for each Frequency point
The centre frequency of wave filter, filter type, the range parameter of filter Q and lifting or decay;
C, the parameter in the speaker frequency response plot correction table that step B is obtained are come to loudspeaker frequency response curve
Rectification module is configured;
D, loudspeaker frequency response curve rectification module good set by step C is individually downloaded to special loudspeaker frequency ring
In answering curve to correct debugging board, and repeat step A, to being tested through the audio amplifier after overcorrection, obtains actual after overcorrection
The audio amplifier frequency response curve for measuring, and contrasted with the target frequency response curve of formulation in step B, if both also have
Difference, then be micro-adjusted to speaker frequency response plot correction table, then bent according to the frequency response of micro-adjustment rear speaker
Parameter in line correction table is configured to loudspeaker frequency response curve rectification module, and step step D is repeated afterwards, until both
It is identical;
Parameter in E, the speaker frequency response plot correction table for completing step D adjustment is come to loudspeaker frequency response curve
Rectification module is configured, then the loudspeaker frequency response curve rectification module that will be set is added to DSP digital signal processing modules
In.
It is as shown in Figure 1 present system structured flowchart.Wherein, LCD display and the rotary encoder with button are use
Family interface, user is to all operation settings of system by turn clockwise operation, the rotate counterclockwise to rotary encoder
Operation or button realize that LCD display is used for the running status and parameter of Display System Function and each functional module, meanwhile,
When user carries out system setting, operation indicating is carried out to user;Audio analog signals input interface receives external audio signal
It is input into and carries out early stage noise reduction and enhanced processing, then signal is delivered to the simulation input port of DSP digital signal processing modules;
DSP digital signal processing modules carry analog/digital conversion and D/A switch, and analog/digital conversion module is converted to audio analog signals
Digital audio and video signals after treatment are converted to analog signal by data signal, D A switch module again, are then sent to sound
Frequency power amplifier rear class is further amplified, and the system is with the related all operations of audio frequency process in DSP Digital Signal Processing moulds
Completed in block;Micro controller module is the control section of whole system, and upon power-up of the system, each module is complete in control system first
Into reset, then DSP program codes and parameter are loaded into DSP by I2C buses, while passing through
Spi bus control LCD display is controlled by flow completion system start step display after system enters normal operating conditions
LCD display system main interface, hereafter starts to receive user's operation setting.
It is illustrated in figure 2 the start step of system:First, system electrification;Micro controller module enters first after system electrification
Row is resetted, and other modules are resetted in control system again after the completion of reset;After all modules complete to reset in waiting system
Control LCD display company Logo, are then loaded into the program code and supplemental characteristic of DSP Digital Signal Processing to DSP data signals
In process chip, but now DSP is in mute state, LCD display product types is controlled after waiting 2S, after and then waiting 2S again
Control LCD display system main interface, while controlling DSP digital signal processing modules to start working, whole system is initially entered just
Normal working condition, microcontroller starts to receive user's operation;Under system main interface state, user can be by the left or to the right
Rotary encoder is rotated to adjust system master volume.
If Fig. 3 is system setting procedure figure:Under system main interface state, the button of rotary encoder is clicked, it will
Enter into system parameter setting state, in Fig. 3 it is listed be in system all functional modules that can be voluntarily set by user and
Its corresponding optional parameters value, and for frequency response curve rectification module, loudness control module, signal pressure limit processing module, power amplifier
Peak power monitoring modular, height cent frequency module etc., are completely to lift the quality of audio amplifier and design, for different sounds
Case and loudspeaker, the concrete function and parameter of module are the fixed optimal cases selected, and are sightless for user, therefore
User can not set these parameters.
After system parameter setting state is entered into, different work(are selected by rotating rotary encoder to the left or to the right
Can menu.Rotary encoder will choose next menu item to the gear of right rotation one, and be kept off to anticlockwise one, can choose
Previous menu, selected menu item can be changed into flip displays, and the corresponding parameter value in menu item right side is then in just
Normal dispaly state.When needing the parameter value to certain function items to modify, rotary encoder is rotated first and chooses the function
The menu label of item, then clicks rotary encoder, then enter into the parameter setting state of the menu item, now, menu item
Label return to normal display state, and by the current parameter value flip displays of the right side menu item, change parameter value keep left it is aobvious
It is shown as being shown centered on, and adds left and right two inverted triangle Direction Signs, when there be optional parameters in current parameter value left side,
Inverted triangle Direction Signs on the left of display, do not show otherwise;When there be optional parameters on current parameter value right side, on the right side of display
Inverted triangle Direction Signs, do not show otherwise.The optional parameters value list of menu item is used and shown successively from left to right or from right to left
The mode shown is selected for user.
When the parameter value of menu item is changed, come into force, that is to say, that what the parameter value of menu item was done is every
Once adjust, a next state that all can be at once to that there should be the corresponding functional module of the menu item in system switches, but simply one
Interim working condition, only when log off parameter setting state when, system can just be preserved to all modifications for being done,
And as the acquiescence working condition of module when starting shooting next time., it is necessary to click rotation after the parameter of certain menu item is set
Turn encoder and exit menu item parameter value setting state, system automatically returns to the selected state of the menu item, can now continue
Parameter value to other function menus is modified, and the method for adjustment of all menu item parameter values is identical.
When system reset is carried out, that is, the working condition of each functional module of system when dispatching from the factory is restored the system to, it is first
First operation rotary encoder chooses system reset menu, then clicks rotary encoder, and system can control LCD display system to answer
Position confirm dialog interface confirmed for user, now, can be selected by rotating rotary encoder to the left or to the right " YES " or
" NO " label, then clicks rotary encoder and notifies the reset operation of system execution system, if that selection is " NO ", system again
It is left intact and is returned directly to the selected state of system reset menu, reset operation terminates;If that selection is " YES ",
Then system starts to perform the operation that resets, and the state of all functional modules in system is returned into state when dispatching from the factory, then automatically
The selected state of system reset menu is returned to, reset operation terminates.
The parameter setting that logs off state., it is necessary to log off parameter setting state simultaneously after completion system parameter setting
System main interface is returned to, is then needed operation rotary encoder first to choose and is exited and preserve menu item, then clicked rotation and compile
Code device, system can first to change after the parameter value of each menu item preserve, preservation automatically returns to system master after terminating
Interface display state.
The implication correspondence of each mark is as follows in the system setup operation flow chart shown in Fig. 3:
Press :Expression clicks rotary encoder
Left :Represent to anticlockwise rotary encoder
Right :Represent to right rotation rotary encoder
Default value:Expression system default parameter value for being used of the menu item when dispatching from the factory or after execution system resets
Start value:Expression is set by default when system is started shooting every time using the parameter value, normal at this in addition
To the working condition on primary system under the influence of the adjustment not of the menu item parameter value during electricity during working condition.
Audio amplifier frequency response curve is corrected:, for the audio amplifier of concrete model, its parameter is by loudspeaker design engineer, sound for the module
Debugging is given jointly for case design engineer and audition engineer, is a fixed functional module, therefore be invisible to user
, user can not be adjusted to its parameter.
It is that the frequency response curve of the audio amplifier of a use DSP audio digital signals treatment prime is being corrected as shown in figure 4 above
Comparing result after preceding and correction, curve 1 is frequency response curve of the audio amplifier bass unit before correction, and curve is changed into after overcorrection
5;Curve 2 is frequency response curve of the audio amplifier voice unit before correction, and curve 6 is changed into after overcorrection.Module has used nine phases
Mutual independent spike wave filter, the parameter of each wave filter is as shown in table 1 below:
Sequence number | Centre frequency:Hz | Quality factor Q values | Lifting or attenuation amplitude: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
Lifting or attenuation amplitude in table, are lifted on the occasion of representing to the signal of the frequency band, and negative value is represented to the frequency
The signal of section is decayed.Frequently, the cut-off frequency of bass is 2700.00Hz to high and low cent, using Lin Kuici-Ruili
(Linkwitz-Riley 48)48 rank wave filters, end gain is 0dB;The initial frequency of high pitch is 2600.00Hz, is equally adopted
With Lin Kuici-Ruili(Linkwitz-Riley 48)48 rank wave filters, end gain is 2.0dB.
Such as optional audio and its corresponding frequency that Fig. 5 is the audio amplifier that a use DSP audio digital signals process prime
Ring curve.Processing method is equally that a certain frequency range of audio signal is lifted or decayed, but uses 10 sections of ginsengs
Number balanced device, constructs a virtual acoustic environments effect, and No. 10 curve MUSIC in figure do not do any treatment to signal, are
The effect state of system default;What No. 11 curve LIVE built is the auditory effect at concert scene;No. 12 curve CLUB build
Be auditory effect in club;No. 13 line SPEECH lines are then the treatment effects when microphone is connect to voice signal.
As Fig. 6 show the audio amplifier of a use DSP audio digital signals treatment prime in installation site difference when institute
The treatment effect contrast for carrying out.Such as under POLE support mounting means, the signal to below 1kHz is lifted, and 60 arrive 500Hz
Between about lift 6dB, weaken for making up the bass loudness that the sound makes an uproar to audio amplifier back side diffusion.
The place of the high and low sound processing module of audio amplifier of a use DSP audio digital signals treatment prime is shown such as Fig. 7
Reason effect, the treatment to bass uses a spike wave filter, and centre frequency is 65.00Hz, quality factor(Q values)It is 1.0, end
End gain is 0;Treatment to high pitch equally uses a spike wave filter, and centre frequency is 11000.00Hz, quality factor(Q
Value)It is 5.0, end gain is 0.The lifting of high and low sound or attenuation range are+10dB~-10dB, and middle curve is
Do not carry out any treatment(0dB)When curve, it is the curve after maximum lift 10dB above to put, and it is maximum attenuation -10dB to transfer
Curve afterwards.High and low sound module is designed to meet personal sense of hearing hobby.
As the super-bass loudspeaker box that Fig. 8 show a use DSP audio digital signals treatment prime is rectified to frequency response curve
The setting of positive effect and subwoofer cut-off frequency.No. 6 curves are original(Without the treatment of DSP audio digital signals)Frequency response
Curve, No. 8 curves are to the effect after frequency response curve correction with the treatment of DSP audio digital signals.Subwoofer cut-off frequency is set
Use a Lin Kuici-Ruili(Linkwitz-Riley 48)48 rank wave filters, the cut-off frequency of No. 10 curves is
150.00Hz, No. 11 cut-off frequencies of curve are 120.00Hz, and No. 12 cut-off frequencies of curve are 100.00Hz, No. 13 curves
Cut-off frequency is 80.00Hz, and No. 14 cut-off frequencies of curve are 60.00Hz, and design parameter is then voluntarily set by user.
Claims (1)
1. a kind of DSP audio digital signals processing system, it is characterised in that:
The DSP audio digital signals processing system includes audio analog signals input interface module, DSP Digital Signal Processing moulds
Block, audio analog signals output interface module, single-chip microcomputer micro-control module, display module and user interface module, it is described
Display module, user interface module and DSP digital signal processing modules are connected with single-chip microcomputer micro-control module signal,
The audio analog signals input interface module and audio analog signals output interface module with DSP Digital Signal Processing moulds
Block signal is connected;
The audio analog signals input interface module is used to connect external audio signal input, signal amplification and improves noise
Than, and the signal after treatment is sent into the analog input interface of DSP digital signal processing modules;
The DSP digital signal processing modules are used to carry out A/D conversions, Digital Signal Processing to the simulated audio signal being input into,
Treatment is changed output and is believed by the high pitch after frequency dividing, bass and subwoofer analog signal to audio frequency simulation by D/A again after terminating
Number output interface module;
The audio analog signals output interface module point high pitch output channel, bass output channel and subwoofer output channel,
High pitch output channel and bass output channel are connected with power amplifier rear class high pitch, bass amplifying circuit respectively, subwoofer output channel
Signal source of the super low sound signal of output directly as super-bass loudspeaker box;
The display module is grasped as user interface for Display System Function, module running status, parameter and user
Indicate;
The all operations of user interface module integrated system are integrated with setting;
The single-chip microcomputer micro-control module is used for the running status of other modules in control system;
The DSP digital signal processing modules include the A/D modular converters of signal connection, loudspeaker frequency response curve correction mould successively
Block and D/A modular converters;
DSP audio digital signals processing methods according to above-mentioned DSP audio digital signals processing system, it is comprised the following steps:
(1), setting loudspeaker frequency response curve corrects parameter and sets other specification by user interface module;
(2), external audio analog signal input audio analog signals input interface module, is then input into by audio analog signals
Interface Module signals amplify and are input to A/D modular converters after improving signal to noise ratio, then change audio by A/D modular converters again
Frequency word signal input is to loudspeaker frequency response curve rectification module;
(3) the loudspeaker frequency response curve correction parameter that, loudspeaker frequency response curve rectification module sets according to step (1) is come to transmission
The audio digital signals for coming over are processed;
(4) audio digital signals after step (3) treatment, are input to D/A modular converters, are then turned by D/A modular converters
Change becomes audio analog signals;
(5) audio analog signals after, step (4) is converted are exported by audio analog signals output interface module;
The setting loudspeaker frequency response curve correction parameter of the step (1) is mainly included the following steps that:
A, first by electroacoustical instrument, the actual frequency response curve and actual phase of audio amplifier are measured in the anechoic room of standard
Position response curve;
B, by actual frequency response curve measured in step A and the mesh formulated by loudspeaker engineer and audio amplifier engineer
Mark frequency response curve is contrasted, and obtains the difference on audio each Frequency point, so as to show that a loudspeaker frequency rings
Answering curve correction table, the correction table includes the number of the frequency correction point being inserted, and filtering needed for each Frequency point
The centre frequency of device, filter type, the range parameter of filter Q and lifting or decay;
C, the parameter in the speaker frequency response plot correction table that step B is obtained are come to the correction of loudspeaker frequency response curve
Module is configured;
D, that loudspeaker frequency response curve rectification module good set by step C is individually downloaded to the special frequency response of loudspeaker is bent
In line correction debugging board, and repeat step A, to being tested through the audio amplifier after overcorrection, obtain actually measured after overcorrection
Audio amplifier frequency response curve, and with step B in formulate target frequency response curve contrasted, if both also have area
Not, then speaker frequency response plot correction table is micro-adjusted, then according to micro-adjustment rear speaker frequency response curve
Parameter in correction table is configured to loudspeaker frequency response curve rectification module, and step step D is repeated afterwards, until both phases
Together;
Parameter in E, the speaker frequency response plot correction table for completing step D adjustment to correct loudspeaker frequency response curve
Module is configured, then the loudspeaker frequency response curve rectification module that will be set is added in DSP digital signal processing modules.
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CN105632511A (en) * | 2015-12-29 | 2016-06-01 | 太仓美宅姬娱乐传媒有限公司 | Sound processing method |
CN105717834A (en) * | 2016-01-29 | 2016-06-29 | 广州市昇博电子科技有限公司 | DSP digital audio processing method |
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CN112887877B (en) * | 2021-01-28 | 2023-09-08 | 歌尔科技有限公司 | Audio parameter setting method and device, electronic equipment and storage medium |
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