CN106358108B - Compensating filter is fitted system, sound equipment compensation system and method - Google Patents
Compensating filter is fitted system, sound equipment compensation system and method Download PDFInfo
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- CN106358108B CN106358108B CN201610791600.3A CN201610791600A CN106358108B CN 106358108 B CN106358108 B CN 106358108B CN 201610791600 A CN201610791600 A CN 201610791600A CN 106358108 B CN106358108 B CN 106358108B
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R1/00—Details of transducers, loudspeakers or microphones
- H04R1/10—Earpieces; Attachments therefor ; Earphones; Monophonic headphones
- H04R1/1091—Details not provided for in groups H04R1/1008 - H04R1/1083
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R1/00—Details of transducers, loudspeakers or microphones
- H04R1/20—Arrangements for obtaining desired frequency or directional characteristics
- H04R1/22—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired frequency characteristic only
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R3/00—Circuits for transducers, loudspeakers or microphones
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R3/00—Circuits for transducers, loudspeakers or microphones
- H04R3/04—Circuits for transducers, loudspeakers or microphones for correcting frequency response
Abstract
The present invention provides a kind of compensating filter fitting system, sound equipment compensation system and method, it includes: goal systems that compensating filter, which is fitted system, including the first transmission function module, the voice signal suitable for that will input exports target sound signal after the first transmission function processing in the first transmission function module；Simulation system is connected with goal systems, including the second transmission function module, and the target sound signal suitable for that will input exports mould analoging sound signal after the second transmission function processing in the second transmission function module；Fitting module is connected with goal systems and simulation system, suitable for target sound signal to be mutually fitted with analoging sound signal, to obtain the parameter for the compensating filter being connected with simulation system input terminal.By obtaining and adding compensating filter, the distortion of simulation system can be eliminated, the reduction target sound signal of high quality may be implemented；It can make a kind of other sound equipments of various types of, the different sound quality weightings of simulation system simulation.
Description
Technical field
The present invention relates to audio areas, more particularly to a kind of compensating filter fitting system, sound equipment compensation system and side
Method.
Background technique
It include various earphones and sound box system in all sound systems, due to factors such as self structures, and from broadcasting
The voice signal put passes through air borne, and space reflection can have various up to the sound that ear is heard compared to primary sound
Loss, delay or distortion.It is big especially for second-rate sound system distortion.
For earphone, currently, either active earphone or Passive earphone are (active, passive whether to refer to earphone
It is self-powered) it can only all issue the sound with own characteristic.Some earphones can restore primary sound well, some earphone basses are muddy
Thickness, some high pitchs are beautiful.But the same earphone only embodies a kind of characteristic now, is unable to satisfy user's various need that it is difficult to cater for all tastes
It asks.It can not accomplish the sound for going simulation B earphone to issue with A earphone.
Summary of the invention
In view of the foregoing deficiencies of prior art, the purpose of the present invention is to provide a kind of fittings of compensating filter is
System, sound equipment compensation system and method, for solving the sound transmission of broadcasting existing for sound equipment existing in the prior art to hearer
There are the problem of various losses, delay or distortion and an earphone or speaker can only body compared to primary sound when ear
The problem of showing a kind of characteristic, being unable to satisfy user's various demands that it is difficult to cater for all tastes.
In order to achieve the above objects and other related objects, the present invention provides a kind of compensating filter fitting system, the benefit
Filter fits system is repaid to include at least:
Goal systems, including the first transmission function module, the voice signal suitable for that will input transmit letter by described first
Target sound signal is exported after the first transmission function processing in digital-to-analogue block；
Simulation system is connected with the goal systems, including the second transmission function module, suitable for the mesh that will be inputted
It marks voice signal and exports simulated target voice signal after the second transmission function processing in the second transmission function module
Analoging sound signal；
Fitting module is connected with the goal systems and the simulation system, be suitable for by the target sound signal with
The analoging sound signal is mutually fitted, to obtain the parameter for the compensating filter being connected with the simulation system input terminal.
Preferably, the compensating filter that first transmission function and second transmission function are FIR or IIR.
Preferably, first transmission function and second transmission function use impulse function method, sef-adapting filter
Approximatioss, Wiener Filter Method or matlab calculating method obtain.
Preferably, the specific method of first transmission function or second transmission function is obtained using impulse function method
Are as follows: playing an impulse function and recording simultaneously is to obtain first transmission function or second transmission function.
Preferably, first transmission function or second transmission function are obtained using sef-adapting filter approximatioss
Method particularly includes: it plays a white noise and records simultaneously, the white noise voice signal and recorded audio signals LMS of foundation broadcasting,
NLMS or RLS algorithm are approached to obtain approximation signal, so that the variance of approximation signal and recorded audio signals is minimum, obtained FIR
Last group of parameter be first transmission function or second transmission function.
Preferably, the specific side of first transmission function or second transmission function is obtained using matlab calculating method
Method are as follows: modeled using ARX, then restrained with recursive Linear-Quadratic Problem mode, first transmission function or institute can be obtained
State the second transmission function.
Preferably, the fitting module controls simulation using FXLMS algorithm, frequency domain phase multiplication, minimum phase method or LQG
The target sound signal is fitted with the voice signal, with obtain being connected with the input terminal of the simulation system
The parameter of compensating filter.
Preferably, the fitting module uses frequency domain phase multiplication by the target sound signal and the analoging sound signal
It is fitted, to obtain the parameter of the compensating filter that is connected with the input terminal of the simulation system method particularly includes: root
It is equal to the principle that frequency domain is multiplied according to convolution, obtains phase to a shock pulse (directly using shock pulse as backoff algorithm)
FFT decomposition should be done, decomposition obtains frequency domain, decomposes in each point of frequency domain divided by the FFT of transmission function (FIR), institute can be obtained
The frequency domain components for stating compensating filter do FFT anti-change, and the parameter of the compensating filter can be obtained.
Preferably, the fitting module uses minimum phase method by the target sound signal and the analoging sound signal
It is fitted, to obtain the parameter of the compensating filter that is connected with the input terminal of the simulation system method particularly includes: false
If the transmission function of simulation system isThe expression formula of the compensating filter isSince simulation system is stablized,It is multinomial of the zero point in unit circle,It cannot be guaranteed that zero point is in unit circle.In order to guarantee the compensating filter
Stablize, the amplitude-frequency response of goal systems and simulation system is constant, and phase frequency meets minimum phase method, does goal systems spectral factorization equation
It asks, makes the multinomial for zero point in unit circle, spectral factorization equation are as follows:
β(z-1) β * (z)=B (z-1)B*(z)
Wherein,
β(z-1)=β0+β1z-1+β2z-2+......+βnβz-nβ；
β * (z)=β0+β1z1+β2z2+......+βnβznβ；
B(z-1)=B0+B1z-1+B2z-2+......+BnBz-nB；
B* (z)=B0+B1z1+B2z2+......+BnBznB
By above formula, the parameter of the compensating filter can be obtained.
Preferably, the fitting module controls simulation for the target sound signal and the simulated sound using LQG
Signal is fitted, to obtain the specific method of the parameter for the compensating filter being connected with the input terminal of the simulation system
Are as follows: on the basis of minimum phase method, it is assumed that the target sound signal first crosses time delay module d, assume that the benefit in this way
The expression formula for repaying filter isIt can be in the hope of by solving Diophantine equation as followsTo obtain institute
State the parameter of compensating filter:
z-dB* (z)=Q (z-1)β*(z)-zL*(z)
Preferably, the energy between the obtained target sound signal of the fitting module and the analoging sound signal
Variance minimum when parameter be the compensating filter parameter.
It preferably, further include time delay module, the output end and the fitting mould of the time delay module and the goal systems
Block is connected, and the target sound signal suitable for issuing the goal systems is transferred to the fitting module after certain time-delay.
It preferably, further include bypass punishment filter, the input terminal and the compensation filter of the bypass punishment filter
The output end of device is connected, and output end is connect with the fitting module, is adapted to filter out limit high frequency and pole in target sound signal
Limit the voice signal of low frequency.
Preferably, the fitting module uses minimum phase method by the target sound signal and the analoging sound signal
It is fitted, to obtain the parameter of the compensating filter that is connected with the input terminal of the simulation system method particularly includes: false
If the transmission function of simulation system isThe transmission function of the bypass punishment filter are as follows:Do spectral factorization equation
It asks, makes the multinomial for zero point in unit circle, spectral factorization equation is；
β(z-1) β * (z)=V (z-1)V*(z)B(z-1)B*(z)+W(z-1)W*(z)A(z-1)A*(z)
By above formula, the parameter of the compensating filter can be obtained.
Preferably, the fitting module controls simulation for the target sound signal and the simulated sound using LQG
Signal is fitted, to obtain the specific method of the parameter for the compensating filter being connected with the input terminal of the simulation system
Are as follows: on the basis of minimum phase method, it is assumed that the target sound signal first crosses time delay module d, assume that the benefit in this way
The expression formula for repaying filter isThe transmission function of the bypass punishment filter are as follows:By such as
Lower solution Diophantine equation can be in the hope ofTo obtain the parameter of the compensating filter:
z-dV(z-1) V* (z) B* (z)=Q (z-1)β*(z)-zL*(z)
The present invention also provides a kind of approximating method of compensating filter, the approximating method of the compensating filter is at least wrapped
It includes:
1) it obtains by treated the target sound signal of the first transmission function in goal systems；
2) it obtains by treated the analoging sound signal of the second transmission function in simulation system；
3) it is fitted according to the target sound signal and the analoging sound signal to obtain the ginseng of compensating filter
Number.
Preferably, in the step 1) and in the step 2), impulse function method is respectively adopted, sef-adapting filter approaches
Method, Wiener Filter Method or matlab calculating method obtain first transmission function or second transmission function.
Preferably, the specific method of first transmission function or second transmission function is obtained using impulse function method
Are as follows: playing an impulse function and recording simultaneously is to obtain first transmission function or second transmission function.
Preferably, first transmission function or second transmission function are obtained using sef-adapting filter approximatioss
Method particularly includes: it plays a white noise and records simultaneously, the white noise voice signal and recorded audio signals LMS of foundation broadcasting,
NLMS or RLS algorithm are approached to obtain approximation signal, so that the variance of approximation signal and recorded audio signals is minimum, obtained FIR
Last group of parameter be first transmission function or second transmission function.
Preferably, the specific side of first transmission function or second transmission function is obtained using matlab calculating method
Method are as follows: modeled using ARX, then restrained with recursive Linear-Quadratic Problem mode, first transmission function or institute can be obtained
State the second transmission function.
Preferably, it in the step 3), is controlled and is simulated using FXLMS algorithm, frequency domain phase multiplication, minimum phase method or LQG
The target sound signal is fitted by method with the voice signal, to obtain being connected with the input terminal of the simulation system
Compensating filter parameter.
Preferably, the target sound signal is fitted with the analoging sound signal using frequency domain phase multiplication, with
Obtain the parameter for the compensating filter being connected with the input terminal of the simulation system method particularly includes: according to convolution etc.
In the principle that frequency domain is multiplied, FFT decomposition is accordingly done to a shock pulse (directly using shock pulse as backoff algorithm),
Decomposition obtains frequency domain, decomposes in each point of frequency domain divided by the FFT of transmission function (FIR), the compensating filter can be obtained
Frequency domain components, do FFT anti-change, the parameter of the compensating filter can be obtained.
Preferably, the target sound signal is fitted with the analoging sound signal using minimum phase method, with
Obtain the parameter for the compensating filter being connected with the input terminal of the simulation system method particularly includes: assuming that simulation system
Transmission function isThe expression formula of the compensating filter isSince simulation system is stablized, A (: it is zero point in list
Multinomial in circle of position,It cannot be guaranteed that zero point is in unit circle.In order to guarantee the stabilization of the compensating filter, target system
System and the amplitude-frequency response of simulation system are constant, and phase frequency meets minimum phase method, does what goal systems spectral factorization equation was asked, makes to be zero
Multinomial of the point in unit circle, spectral factorization equation are as follows:
β(z-1) β * (z)=B (z-1)B*(z)
Wherein,
By above formula, the parameter of the compensating filter can be obtained.
Preferably, the target sound signal is fitted with the analoging sound signal using LQG control simulation,
To obtain the parameter for the compensating filter being connected with the input terminal of the simulation system method particularly includes: in minimum phase method
On the basis of, it is assumed that the target sound signal first crosses time delay module d, assume that the expression formula of the compensating filter in this way
ForIt can be in the hope of by solving Diophantine equation as followsTo obtain the ginseng of the compensating filter
Number:
z-dB* (z)=Q (z-1)β*(z)-zL*(z)
Preferably, the target sound signal is fitted with the analoging sound signal using minimum phase method, with
Obtain the parameter for the compensating filter being connected with the input terminal of the simulation system method particularly includes: assuming that simulation system
Transmission function isThe transmission function of the bypass punishment filter are as follows:It does what spectral factorization equation was asked, makes for zero point
Multinomial in unit circle, spectral factorization equation are；
β(z-1) β * (z)=V (z-1)V*(z)B(z-1)B*(z)+W(z-1)W*(z)A(z-1)A*(z)
By above formula, the parameter of the compensating filter can be obtained.
Preferably, the target sound signal is fitted with the analoging sound signal using LQG control simulation,
To obtain the parameter for the compensating filter being connected with the input terminal of the simulation system method particularly includes: in minimum phase method
On the basis of, it is assumed that the target sound signal first crosses time delay module d, assume that the expression formula of the compensating filter in this way
ForThe transmission function of the bypass punishment filter are as follows:By solving Diophantine equation as follows
It can be in the hope ofTo obtain the parameter of the compensating filter:
z-dV(z-1) V* (z) B* (z)=Q (z-1)β*(z)-zL*(z)
The present invention also provides a kind of sound equipment compensation system, the Sound Imitation System is included at least:
Simulation system, including the second transmission function module, the target sound signal suitable for providing goal systems pass through institute
State the analoging sound signal that simulated target voice signal is exported after the second transmission function in the second transmission function module is handled；
Compensating filter, including input terminal and output end；The input of the compensating filter is connected with the goal systems
It connects, the output end of the compensating filter is connected with the input terminal of the simulation system, is suitable for the input simulation system
Voice signal handled so that the variance of the energy of the analoging sound signal and the target sound signal is minimum；
The parameter of the compensating filter is obtained as the fitting system of the compensating filter as described in above-mentioned either a program.
Preferably, the simulation system and the goal systems are earphone or speaker.
Preferably, stating simulation system is Passive earphone or passive loudspeaker box, the input terminal of the compensating filter and the mesh
Mark system is connected, and output end is connected with the input terminal of the simulation system.
Preferably, the simulation system is active earphone or active audio amplifier, and the compensating filter is located at simulation system
System is internal.
The present invention also provides a kind of sound simulation method, the sound simulation method is included at least:
1) compensating filter is provided, the parameter of the compensating filter is as the compensation filter as described in above-mentioned either a program
Device is fitted system and obtains；
2) target sound signal the compensating filter is inputted to handle；
3) by the compensating filter, treated that voice signal is sent in simulation system is played.
As described above, compensating filter fitting system, sound equipment compensation system and method for the invention, have below beneficial to effect
Fruit: by adding compensating filter, can eliminate the distortion of simulation system, and the reduction target sound letter of high quality may be implemented
Number；It can make a kind of other sound equipments of various types of, the different sound quality weightings of simulation system simulation；By adding bypass punishment
Filter can remove the sound frequency range that simulation system can not issue, and can improve sound in the case where identical output energy
Amount, to reduce unnecessary capacity loss, and protects simulation system not to be destroyed in the limiting case.
Detailed description of the invention
Fig. 1 is shown as the block diagram of the compensating filter provided in the embodiment of the present invention one fitting system.
Fig. 2 is shown as the fitting module of the compensating filter provided in the embodiment of the present invention one fitting system using FXLMS
The block diagram of fitting algorithm fitting.
Fig. 3 is shown as the fitting module in the compensating filter fitting system provided in the embodiment of the present invention one using minimum
The shock response figure of phase method fitting, wherein dotted line is the output of compensating filter, and solid line is the voice signal finally heard.
Fig. 4 is shown as the fitting module in the compensating filter fitting system provided in the embodiment of the present invention one using LQG
The shock response figure of fitting, wherein dotted line is the output of compensating filter, and solid line is the sound finally heard.
Fig. 5, which is shown as the another kind provided in the embodiment of the present invention one, has the compensating filter of bypass punishment filter quasi-
The block diagram of collaboration system.
What Fig. 6 was shown as providing in the embodiment of the present invention one has the compensating filter fitting system of bypass punishment filter
In shock response figure.
Fig. 7 is shown as the flow chart of the approximating method of compensating filter provided by Embodiment 2 of the present invention.
Fig. 8 is shown as the block diagram of the Sound Imitation System provided in the embodiment of the present invention three.
Fig. 9 is shown as the flow chart of the sound simulation method of the offer of the embodiment of the present invention four.
Component label instructions
1 simulation system
2 fitting modules
21 adders
22 fitting units
3 compensating filters
4 time delay modules
5 bypass punishment filters
Specific embodiment
Illustrate embodiments of the present invention below by way of specific specific example, those skilled in the art can be by this specification
Other advantages and efficacy of the present invention can be easily understood for disclosed content.The present invention can also pass through in addition different specific realities
The mode of applying is embodied or practiced, the various details in this specification can also based on different viewpoints and application, without departing from
Various modifications or alterations are carried out under spirit of the invention.
Fig. 1 is please referred to Fig. 9.It should be noted that diagram provided in the present embodiment only illustrates this in a schematic way
The basic conception of invention, though only show in diagram with related component in the present invention rather than package count when according to actual implementation
Mesh, shape and size are drawn, when actual implementation kenel, quantity and the ratio of each component can arbitrarily change for one kind, and its
Assembly layout kenel may also be increasingly complex.
Embodiment one
Referring to Fig. 1, the compensating filter fitting system includes at least: goal systems (not shown), the target packet
The first transmission function module (not shown) is included, the voice signal that the goal systems is suitable for input is by the first transmitting letter
Target sound signal w (k) is exported after the first transmission function processing in digital-to-analogue block；Simulation system 1, the simulation system 1 and institute
It states goal systems to be connected, the simulation system 1 includes the second transmission function module (not shown), suitable for the mesh that will be inputted
Mark voice signal w (k)) simulated target sound is exported after the second transmission function processing in the second transmission function module
The analoging sound signal ym (k) of signal)；Fitting module 2, the fitting module 2 and the goal systems and the simulation system 1
It is connected, is suitable for for the target sound signal being mutually fitted with the analoging sound signal ym (k), obtains being with the simulation
The parameter for the compensating filter 2 that system input terminal is connected.
As an example, the compensating filter that first transmission function and second transmission function are FIR or IIR.
As an example, between the obtained target sound signal of the fitting module 2 and the analoging sound signal
Parameter when the variance minimum of energy is the parameter of the compensating filter 3.
As an example, as shown in Figure 1, compensating filter fitting system further includes time delay module 4, the time delay module
4 are connected with the output end of the goal systems and the fitting module 2, suitable for the target sound for issuing the goal systems
Target sound signal yref (k) to be delayed after of the signal w (k) after certain time-delay, then by the target sound signal
Yref (k) is sent to the fitting module 2.
As an example, first transmission function and second transmission function method particularly includes: assume first that input
Electric signal to the goal systems is x (0), x (1) ... the sequence of x (n), and human ear hears (or mic (microphone) recording obtain)
Be (0) y, y (1) ... the sequence of y (n), the relational expression between them can indicate are as follows:
A0y (n)+a1y (n-1) ...+ajy (n-j)=b0x (n)+b1x (n-1) ...+bk (n or
Wherein, the as transmission function of target sound signal or the analoging sound signal.
As an example, first transmission function and second transmission function use impulse function method, adaptive-filtering
Device approximatioss, Wiener Filter Method or matlab calculating method obtain.
As an example, obtaining the specific side of first transmission function or second transmission function using impulse function method
Method are as follows: playing an impulse function and recording simultaneously is to obtain first transmission function or second transmission function.
As an example, obtaining first transmission function or second transmission function using sef-adapting filter approximatioss
Method particularly includes: it plays a white noise and records simultaneously, voice signal will be played and be denoted as x (n), recorded audio signals are denoted as d
(n), it is approached to obtain approximation signal y (n) with LMS, NLMS or RLS algorithm according to x (n) and d (n), so that y (n)-d (n)
Variance is minimum, and last group of parameter of obtained FIR is first transmission function or second transmission function.
As an example, obtaining the specific of first transmission function or second transmission function using matlab calculating method
Method are as follows: modeled using ARX, then restrained with recursive Linear-Quadratic Problem mode, can be obtained first transmission function or
Second transmission function carries out modeling using the tfest of matlab and recurrence obtains first transmission function or described
Second transmission function.
As an example, firstly, by the difference of the analoging sound signal ym (k) and the target sound signal yref (k)
Ability be denoted as E (), E () is smaller, and the analoging sound signal that the simulation system 1 exports gets over phase with the target sound signal
Closely.In order to keep E () minimum, the fitting module 2 can be using FXLMS algorithm, frequency domain phase multiplication, minimum phase method or LQG control
The transmission function of the transmission function of the target sound signal and the analoging sound signal is fitted by simulation method processed, with
Obtain the parameter for the compensating filter 3 being connected with the input terminal of the simulation system 1.
As an example, using FXLMS algorithm by the transmission function of the target sound signal and the analoging sound signal
Transmission function be fitted method particularly includes: as shown in Fig. 2, the estimation function of simulation system 1 described in Fig. 2 is denoted as A ',
Goal systems (not shown) input white noise x (n), white noise x (n) pass through the voice signal y after the compensating filter 3
(n) by the simulation system 1 to the fitting module 2, meanwhile, white noise is obtained into described prolong by the time delay module 4
When the target sound signal d (n) that exports of module 4, analoging sound signal and target sound signal elder generation are carried out via adder 21
Minimum variance operation is carried out by FXLMS algorithm by fitting unit 22 after plus and minus calculation, obtains the ginseng of the compensating filter 3
Number.
As an example, using frequency domain phase multiplication by the transmission function of the target sound signal and the analoging sound signal
Transmission function be fitted method particularly includes: in Fig. 1, the principle that frequency domain is multiplied is equal to according to convolution, to a punching
FFT decomposition is accordingly done in shock pulse (directly using shock pulse as backoff algorithm), and decomposition obtains frequency domain, in each of frequency domain
A point is decomposed divided by the FFT of transmission function (FIR), and the frequency domain components of the compensating filter 3 can be obtained, do FFT anti-change,
The compensating filter 3 can be obtained.
As an example, using minimum phase method by the transmission function of the target sound signal and the analoging sound signal
Transmission function be fitted method particularly includes: assuming that the transmission function for the analoging sound signal that the simulation system 1 exports
ForThe simulation system 1 simulates the transmission function of the target sound signal, one can consider that the compensating filter 3
Expression formula isSince the simulation system 1 is stablized, since A (is multinomial of the zero point in unit circle, B (cannot
Guarantee zero point in unit circle.In order to guarantee the stabilization of the compensating filter 3, the goal systems and the simulation system 1
Amplitude-frequency response it is constant, phase frequency meets minimum phase method, we do goal systems spectral factorization equation and acquire
Make the multinomial for zero point in unit circle.Above in expression formula,
β(z-1)=β0+β1z-1+β2z-2+......+βnβz-nβ；
β * (z)=β0+β1z1+β2z2+......+βnβznβ；
B(z-1)=B0+B1z-1+B2z-2+......+BnBz-nB；
B* (z)=B0+B1z1+B2z2+......+BnBznB
The expression formula R of the i.e. described compensating filter 3 can be substituted forFig. 3 is to be rushed using what this method was fitted
Response diagram is hit, from the figure 3, it may be seen that the compensating filter 3 of this method fitting is to the voice signal for being input to the simulation system 1
After processing, there is preferable effect.
As an example, the fitting module controls simulation for the target sound signal and the simulated sound using LQG
Sound signal is fitted, to obtain the specific method of the parameter for the compensating filter being connected with the input terminal of the simulation system
Are as follows: target sound signal first excessively a time delay module 4 is assumed on the basis of minimum phase method, then as shown in Figure 1, assuming in this way
The expression formula of the compensating filter 3 isTo ask minimum E (), by solving Diophantine equation
z-dB* (z)=Q (z-1)β*(z)-zL*(z)
Wherein, L* (z) be goal systems Diophantine equation another solution, can in the hope of Q (to obtain R.This method
Than the above method good place be Q (be a pre- oscillation picture, can be to the analoging sound signal that the simulation system 1 exports
Transmission function isPre- starting of oscillation is carried out, better phase-frequency response is obtained.Fig. 4 is the shock response figure being fitted using this method, by
Fig. 4 it is found that this method fitting the compensating filter 3 to the sound signal processing of the simulation system 1 is input to after, tool
There is preferable effect.
As an example, referring to Fig. 5, the compensating filter fitting system further include a bypass punishment filter 5, it is described
The input terminal of bypass punishment filter 5 is connected with the output end of the compensating filter 3, output end and the fitting module 2
Connection, is adapted to filter out the voice signal of limit high frequency (17KHz or more) and limit low frequency (50Hz or less) in target sound signal.
Fig. 6 is the shock response figure in this implementation, wherein 2 rank 50Hz low passes and 4 ranks are respectively added in black lines and grey lines
Shock response after the punishment filter of 17000Hz high pass.Filter 5 is punished by adding the bypass, can remove mould
The sound frequency range that quasi- system 1 can not issue, can improve volume in the case where identical output energy, unnecessary to reduce
Capacity loss, and simulation system 1 is protected not to be destroyed in the limiting case.
As an example, add after bypass punishment filter 5, the fitting module 2 will be through using minimum phase method
Cross the target sound signal yref (t) of the time delay module 4, by the target sound of the bypass punishment filter 5
Sound signal yp (t) is fitted with the analoging sound signal ym (k), makes ENERGY E | | yref (t)-ym (+E | | yp (it is minimum, with
Obtain the parameter for the compensating filter 3 being connected with the input terminal of the simulation system 1 method particularly includes: assuming that simulation system
1 transmission function isThe transmission function of the bypass punishment filter 5 are as follows:Do what spectral factorization equation was asked, make for
Multinomial of the zero point in unit circle, spectral factorization equation are；
β(z-1) β * (z)=V (z-1)V*(z)B(z-1)B*(z)+W(z-1)W*(z)A(z-1)A*(z)
z-dV(z-1) V* (z) B* (z)=Q (z-1)β*(z)-zL*(z)
The present invention can eliminate the distortion of simulation system 1, high quality may be implemented by adding the compensating filter 4
Reduction target sound signal.
Embodiment two
Referring to Fig. 7, the present invention also provides a kind of approximating method of compensating filter, the fitting side of the compensating filter
Method includes at least:
1) it obtains by treated the target sound signal of the first transmission function in goal systems；
2) it obtains by treated the analoging sound signal of the second transmission function in simulation system；
3) it is fitted according to the target sound signal and the analoging sound signal to obtain the ginseng of compensating filter
Number.
As an example, in the step 1), it is first assumed that the electric signal for being input to the goal systems is x (0), x
(1) ... the sequence of x (n), what human ear heard (or mic (microphone) recording obtain) is (0) y, y (1) ... the sequence of y (n), it
Between relational expression can indicate are as follows:
A0y (n)+a1y (n-1) ...+ajy (n-j)=b0x (n)+b1x (n-1) ...+bk (n or
Wherein, the as transmission function of target sound signal or the analoging sound signal.
As an example, can be approached using impulse function method, sef-adapting filter in the step 1) and the step 2)
Method, Wiener Filter Method or matlab calculating method obtain first transmission function or second transmission function.
As an example, obtaining the specific side of first transmission function or second transmission function using impulse function method
Method are as follows: playing an impulse function and recording simultaneously is to obtain first transmission function or second transmission function.
As an example, obtaining first transmission function or second transmission function using sef-adapting filter approximatioss
Method particularly includes: it plays a white noise and records simultaneously, voice signal will be played and be denoted as x (n), recorded audio signals are denoted as d
(n), it is approached to obtain approximation signal y (n) with LMS, NLMS or RLS algorithm according to x (n) and d (n), so that y (n)-d (n)
Variance is minimum, and last group of parameter of obtained FIR is first transmission function or second transmission function.
As an example, obtaining the specific of first transmission function or second transmission function using matlab calculating method
Method are as follows: modeled using ARX, then restrained with recursive Linear-Quadratic Problem mode, can be obtained first transmission function or
Second transmission function carries out modeling using the tfest of matlab and recurrence obtains first transmission function or described
Second transmission function.
As an example, controlling mould using FXLMS algorithm, frequency domain phase multiplication, minimum phase method or LQG in the step 3)
The target sound signal is fitted by quasi- method with the voice signal, to obtain being connected with the input terminal of the simulation system
The parameter of the compensating filter connect.
As an example, using FXLMS algorithm by the transmission function of the target sound signal and the analoging sound signal
Transmission function be fitted method particularly includes: as shown in Fig. 2, the estimation function of simulation system 1 described in Fig. 2 is denoted as A ',
Goal systems (not shown) input white noise x (n), white noise x (n) pass through the voice signal y after the compensating filter 3
(n) by the simulation system 1 to the fitting module 2, meanwhile, white noise is obtained into described prolong by the time delay module 4
When the target sound signal d (n) that exports of module 4, analoging sound signal and target sound signal elder generation are carried out via adder 21
Minimum variance operation is carried out by FXLMS algorithm by fitting unit 22 after plus and minus calculation, obtains the ginseng of the compensating filter 3
Number.
As an example, using frequency domain phase multiplication by the transmission function of the target sound signal and the analoging sound signal
Transmission function be fitted method particularly includes: in Fig. 1, the principle that frequency domain is multiplied is equal to according to convolution, to a punching
FFT decomposition is accordingly done in shock pulse (directly using shock pulse as backoff algorithm), and decomposition obtains frequency domain, in each of frequency domain
A point is decomposed divided by the FFT of transmission function (FIR), and the frequency domain components of the compensating filter 3 can be obtained, do FFT anti-change,
The compensating filter 3 can be obtained.
As an example, using minimum phase method by the transmission function of the target sound signal and the analoging sound signal
Transmission function be fitted method particularly includes: assuming that the transmission function for the analoging sound signal that the simulation system 1 exports
ForThe simulation system 1 simulates the transmission function of the target sound signal, one can consider that the compensating filter 3
Expression formula beSince the simulation system 1 is stablized, since A (is multinomial of the zero point in unit circle, B is (no
It can guarantee zero point in unit circle.In order to guarantee the stabilization of the compensating filter 3, the goal systems and the simulation system
1 amplitude-frequency response is constant, and phase frequency meets minimum phase method, we do goal systems spectral factorization equation and acquire
Make the multinomial for zero point in unit circle.Above in expression formula,
β(z-1)=β0+β1z-1+β2z-2+......+βnβz-nβ；
β * (z)=β0+β1z1+β2z2+......+βnβznβ；
B(z-1)=B0+B1z-1+B2z-2+......+BnBz-nB；
B* (z)=B0+B1z1+B2z2+......+BnBznB
The expression formula R of the i.e. described compensating filter 3 can be substituted forFig. 3 is to be rushed using what this method was fitted
Response diagram is hit, from the figure 3, it may be seen that the compensating filter 3 of this method fitting is to the voice signal for being input to the simulation system 1
After processing, there is preferable effect.
As an example, the fitting module controls simulation for the target sound signal and the simulated sound using LQG
Sound signal is fitted, to obtain the specific method of the parameter for the compensating filter being connected with the input terminal of the simulation system
Are as follows: target sound signal first excessively a time delay module 4 is assumed on the basis of minimum phase method, then as shown in Figure 1, assuming in this way
The expression formula of the compensating filter 3 isTo ask minimum E (), by solving Diophantine equation
Can in the hope of Q (to obtain R.This method place better than the above method be Q (be a pre- oscillation picture, can be with
The transmission function of analoging sound signal to the simulation system 1 output isPre- starting of oscillation is carried out, better phase-frequency response is obtained.
Fig. 4 is the shock response figure being fitted using this method, and as shown in Figure 4, the compensating filter 3 of this method fitting is to being input to
After the sound signal processing of the simulation system 1, there is preferable effect.
As an example, add after bypass punishment filter 5, the fitting module 2 will be through using minimum phase method
Cross the target sound signal yref (t) of the time delay module 4, by the target sound of the bypass punishment filter 5
Sound signal yp (t) is fitted with the analoging sound signal ym (k), makes ENERGY E | | yref (t)-ym (+E | | yp (it is minimum, with
Obtain the parameter for the compensating filter 3 being connected with the input terminal of the simulation system 1 method particularly includes: assuming that simulation system
1 transmission function isThe transmission function of the bypass punishment filter 5 are as follows:Do what spectral factorization equation was asked, make for
Multinomial of the zero point in unit circle, spectral factorization equation are；
z-dV(z-1) V* (z) B* (z)=Q (z-1)β*(z)-zL*(z)
Embodiment three
Referring to Fig. 8, the Sound Imitation System includes at least the present invention also provides a kind of sound equipment compensation system: simulation
System 1, the target sound signal that the simulation system 1 is suitable for providing according to goal systems generate analoging sound signal, the mould
Onomatopoeia sound signal is the voice signal for simulating the target sound signal；Compensating filter 3, the compensating filter 3 wrap
Include input terminal and output end；The input of the compensating filter 3 is connected with the goal systems, the compensating filter 3
Output end is connected with the input terminal of the simulation system 1, at the voice signal to the input simulation system 1
Reason, so that the variance of the energy of the analoging sound signal and the target sound signal is minimum；The compensating filter 3
Parameter compensating filter as described in embodiment one is fitted system and obtains.
As an example, the simulation system 1 and the goal systems can be earphone or speaker.
As an example, the simulation system 1 can be Passive earphone or speaker, the compensating filter 3 is located at the mould
The outside of quasi- system 1, the input terminal of the compensating filter 3 are connected with the goal systems, and output end and the simulation are
The input terminal of system 1 is connected.
As an example, the simulation system 1 can be active loundspeakers, the compensating filter 3 is located at the simulation system 1
It is internal.
Example IV
Referring to Fig. 9, the sound simulation method includes at least the present invention also provides a kind of sound simulation method:
1) compensating filter, parameter compensating filter as described in embodiment one fitting of the compensating filter are provided
System obtains；
2) target sound signal the compensating filter is inputted to handle；
3) by the compensating filter, treated that voice signal is sent in simulation system is played
As an example, the simulation system can be passive sound, it can be Passive earphone or speaker, the compensation filter
Wave device is located at the outside of the simulation system, and the input terminal of the compensating filter is connected with the goal systems, output end
It is connected with the input terminal of the simulation system.
As an example, the simulation system can be active loundspeakers, it can be active earphone or speaker, the compensation filter
Wave device is located inside the simulation system.
In conclusion compensating filter fitting system, sound equipment compensation system and method for the invention, the compensating filter
Fitting system includes at least: goal systems, including the first transmission function module, suitable for the voice signal that will input by described the
Target sound signal is exported after the first transmission function processing in one transmission function module；Simulation system, with the goal systems
It is connected, including the second transmission function module, the target sound signal suitable for that will input passes through second transmission function
The analoging sound signal of simulated target voice signal is exported after the second transmission function processing in module；Fitting module, and it is described
Goal systems and the simulation system are connected, suitable for the target sound signal is mutually fitted with the analoging sound signal,
To obtain the parameter for the compensating filter being connected with the simulation system input terminal.By adding compensating filter, can disappear
Except the distortion of simulation system, the reduction target sound signal of high quality may be implemented；A kind of simulation system simulation can be made various
Other sound equipments that different type, different sound quality are laid particular stress on.
The above-described embodiments merely illustrate the principles and effects of the present invention, and is not intended to limit the present invention.It is any ripe
The personage for knowing this technology all without departing from the spirit and scope of the present invention, carries out modifications and changes to above-described embodiment.Cause
This, institute is complete without departing from the spirit and technical ideas disclosed in the present invention by those of ordinary skill in the art such as
At all equivalent modifications or change, should be covered by the claims of the present invention.
Claims (29)
1. a kind of compensating filter is fitted system, which is characterized in that include at least:
Goal systems, including the first transmission function module, the voice signal suitable for that will input pass through the first transmission function mould
Target sound signal is exported after the first transmission function processing in block；
Simulation system is connected with the goal systems, including the second transmission function module, suitable for the target sound that will be inputted
Sound signal exports the mould of simulated target voice signal after the second transmission function processing in the second transmission function module
Onomatopoeia sound signal；
Fitting module is connected with the goal systems and the simulation system, be suitable for by the target sound signal with it is described
Analoging sound signal is mutually fitted, to obtain the parameter for the compensating filter being connected with the simulation system input terminal.
2. compensating filter according to claim 1 is fitted system, it is characterised in that: first transmission function and described
Second transmission function is the compensating filter of FIR or IIR.
3. compensating filter according to claim 1 is fitted system, it is characterised in that: first transmission function and described
Second transmission function is obtained using impulse function method, sef-adapting filter approximatioss, Wiener Filter Method or matlab calculating method.
4. compensating filter according to claim 3 is fitted system, it is characterised in that: using described in the acquisition of impulse function method
First transmission function or second transmission function method particularly includes: play an impulse function and record simultaneously and obtains institute
State the first transmission function or second transmission function.
5. compensating filter according to claim 3 is fitted system, it is characterised in that: use sef-adapting filter approximatioss
Obtain first transmission function or second transmission function method particularly includes: it plays a white noise and records simultaneously,
It is approached to obtain approximation signal with LMS, NLMS or RLS algorithm according to the white noise voice signal and recorded audio signals played, be made
Approximation signal and recorded audio signals variance it is minimum, last group of parameter of obtained FIR be first transmission function or
Second transmission function.
6. compensating filter according to claim 3 is fitted system, it is characterised in that: obtain institute using matlab calculating method
State the first transmission function or second transmission function method particularly includes: model using ARX, then with recursive linear quadratic
Type mode restrains, and first transmission function or second transmission function can be obtained.
7. compensating filter according to claim 1 is fitted system, it is characterised in that: the fitting module uses FXLMS
Algorithm, frequency domain phase multiplication, minimum phase method or LQG control simulation carry out the target sound signal and the voice signal
Fitting, to obtain the parameter for the compensating filter being connected with the input terminal of the simulation system.
8. compensating filter according to claim 7 is fitted system, it is characterised in that: the fitting module uses frequency domain phase
The target sound signal is fitted by multiplication with the analoging sound signal, to obtain the input terminal with the simulation system
The parameter for the compensating filter being connected method particularly includes: the principle that frequency domain is multiplied is equal to according to convolution, a punching is provided
Shock pulse obtains response and does FFT decomposition, and decomposition obtains frequency domain, decomposes in each point of frequency domain divided by the FFT of transmission function
The frequency domain components of the compensating filter are obtained, FFT anti-change is done, the parameter of the compensating filter can be obtained.
9. compensating filter according to claim 7 is fitted system, it is characterised in that: the fitting module is using minimum phase
The target sound signal is fitted by position method with the analoging sound signal, to obtain the input terminal with the simulation system
The parameter for the compensating filter being connected method particularly includes: assuming that the transmission function of simulation system isThe compensation
The expression formula of filter isSince simulation system is stablized, A (z-1) it is multinomial of the zero point in unit circle, B
(z-1) cannot be guaranteed zero point in unit circle；In order to guarantee the stabilization of the compensating filter, goal systems and simulation system
Amplitude-frequency response is constant, and phase frequency meets minimum phase method, is the β that goal systems spectral factorization equation is asked, makes β zero point in unit circle
Multinomial, spectral factorization equation are as follows:
β(z-1) β * (z)=B (z-1)β*(z)β(z-1) β * (z)=B (z-1)B*(z)
Wherein,
β(z-1)=β0+β1z-1+β2z-2+......+βnβz-nβ；
β * (z)=β0+β1z1+β2z2+......+βnβznβ；B(z-1)=B0+B1z-1+B2z-2+......+BnBz-nB；
B* (z)=B0+B1z1+B2z2+......+BnBznB
By above formula, the parameter of the compensating filter can be obtained.
10. compensating filter according to claim 9 is fitted system, it is characterised in that: the fitting module is controlled using LQG
The target sound signal is fitted by simulation processed with the analoging sound signal, defeated with the simulation system to obtain
Enter the parameter for the compensating filter that end is connected method particularly includes: on the basis of minimum phase method, it is assumed that the target sound
Sound signal first crosses time delay module d, assume that the expression formula of the compensating filter is in this wayBy such as
Lower solution Diophantine equation can be in the hope of Q (q-1), to obtain the parameter of the compensating filter:
z-dB* (z)=Q (z-1)β*(z)-zL*(z)z-dB* (z)=Q (z-1)β*(z)-zL*(z)。
11. compensating filter according to claim 1 is fitted system, it is characterised in that: the institute that the fitting module obtains
Parameter when stating the variance minimum of the energy between target sound signal and the analoging sound signal is the compensation filter
The parameter of device.
12. compensating filter according to claim 1 is fitted system, it is characterised in that: it further include time delay module, it is described to prolong
When module be connected with the output end of the goal systems and the fitting module, suitable for the target for issuing the goal systems
Voice signal is transferred to the fitting module after certain time-delay.
13. compensating filter according to claim 1 is fitted system, it is characterised in that: it further include bypass punishment filter,
The input terminal of the bypass punishment filter is connected with the output end of the compensating filter, output end and the fitting module
Connection, is adapted to filter out the voice signal of limit high frequency and limit low frequency in target sound signal.
14. compensating filter according to claim 13 is fitted system, it is characterised in that: the fitting module is using minimum
The target sound signal is fitted by phase method with the analoging sound signal, to obtain the input with the simulation system
Hold the parameter for the compensating filter being connected method particularly includes: assuming that the transmission function of simulation system isThe side
The transmission function of road punishment filter are as follows:It is the β that spectral factorization equation is asked, keeps β zero point multinomial in unit circle
Formula, spectral factorization equation are；
β(z-1) β * (z)=V (z-1)V*(z)B(z-1)B*(z)+W(z-1)W*(z)A(z-1)A*(z)
β(z-1) β * (z)=V (z-1)V*(z)B(z-1)B*(z)+W(z-1)W*(z)A(z-1)A*(z)
By above formula, the parameter of the compensating filter can be obtained.
15. compensating filter according to claim 13 is fitted system, it is characterised in that: the fitting module uses LQG
The target sound signal is fitted by control simulation with the analoging sound signal, to obtain and the simulation system
The parameter for the compensating filter that input terminal is connected method particularly includes: on the basis of minimum phase method, it is assumed that the target
Voice signal first crosses time delay module d, assume that the expression formula of the compensating filter is in this wayIt is described
The transmission function of bypass punishment filter are as follows:It can be in the hope of Q (q by solving Diophantine equation as follows-1), thus
To the parameter of the compensating filter:
z-dV(z-1) V* (z) B* (z)=Q (z-1)β*(z)-zL*(z)
z-dV(z-1) V* (z) B* (z)=Q (z-1)β*(z)-zL*(z)。
16. a kind of compensating filter based on the fitting system of compensating filter described in any one of claims 1 to 15 is quasi-
Conjunction method, which is characterized in that the approximating method of the compensating filter includes at least:
1) it is obtained using impulse function method, sef-adapting filter approximatioss, Wiener Filter Method or matlab calculating method and passes through target
First transmission function treated target sound signal in system；
2) it is obtained using impulse function method, sef-adapting filter approximatioss, Wiener Filter Method or matlab calculating method by simulation
Second transmission function treated analoging sound signal in system；
3) using FXLMS algorithm, frequency domain phase multiplication, minimum phase method or LQG control simulation by the target sound signal with
The voice signal is fitted to obtain the parameter of compensating filter.
17. the approximating method of compensating filter according to claim 16, it is characterised in that: obtained using impulse function method
First transmission function or second transmission function method particularly includes: play an impulse function and record simultaneously to obtain the final product
To first transmission function or second transmission function.
18. the approximating method of compensating filter according to claim 16, it is characterised in that: forced using sef-adapting filter
Nearly method obtains first transmission function or second transmission function method particularly includes: plays a white noise and records simultaneously
Sound is approached to obtain approximation signal according to the white noise voice signal and recorded audio signals played with LMS, NLMS or RLS algorithm,
So that the variance of approximation signal and recorded audio signals is minimum, last group of parameter of obtained FIR is first transmission function
Or second transmission function.
19. the approximating method of compensating filter according to claim 16, it is characterised in that: obtained using matlab calculating method
Take first transmission function or second transmission function method particularly includes: model using ARX, then with recursive linear
Quadratic form mode restrains, and first transmission function or second transmission function can be obtained.
20. the approximating method of compensating filter according to claim 16, it is characterised in that: use frequency domain phase multiplication by institute
It states target sound signal to be fitted with the analoging sound signal, with obtain being connected with the input terminal of the simulation system
The parameter of compensating filter method particularly includes: the principle that frequency domain is multiplied is equal to according to convolution, a shock pulse is provided and is obtained
FFT decomposition is done to response, decomposition obtains frequency domain, decomposes, can be obtained divided by the FFT of transmission function (FIR) in each point of frequency domain
To the frequency domain components of the compensating filter, FFT anti-change is done, the parameter of the compensating filter can be obtained.
21. the approximating method of compensating filter according to claim 16, it is characterised in that: use minimum phase method by institute
It states target sound signal to be fitted with the analoging sound signal, with obtain being connected with the input terminal of the simulation system
The parameter of compensating filter method particularly includes: assuming that the transmission function of simulation system isThe compensating filter
Expression formula isSince simulation system is stablized, A (z-1) it is multinomial of the zero point in unit circle, B (z-1) cannot protect
Zero point is demonstrate,proved in unit circle；In order to guarantee the stabilization of the compensating filter, the amplitude-frequency response of goal systems and simulation system is not
Become, phase frequency meets minimum phase method, does what goal systems spectral factorization equation was asked, makes the multinomial for zero point in unit circle, composes
Decompose equation are as follows:
β(z-1) β * (z)=B (z-1)β*(z)β(z-1) β * (z)=B (z-1)B*(z)
Wherein,
β(z-1)=β0+β1z-1+β2z-2+......+βnβz-nβ；
β * (z)=β0+β1z1+β2z2+......+βnβznβ；B(z-1)=B0+B1z-1+B2z-2+......+BnBz-nB；
B* (z)=B0+B1z1+B2z2+......+BnBznB
By above formula, the parameter of the compensating filter can be obtained.
22. the approximating method of compensating filter according to claim 16, it is characterised in that: control simulation using LQG
The target sound signal is fitted with the analoging sound signal, to obtain being connected with the input terminal of the simulation system
The parameter of the compensating filter connect method particularly includes: on the basis of minimum phase method, it is assumed that the target sound signal is first
Time delay module d is crossed, assume that the expression formula of the compensating filter is in this wayIt is lost by solving as follows
Kind figure equation can be in the hope of Q (q-1), to obtain the parameter of the compensating filter:
z-dB* (z)=Q (z-1)β*(z)-zL*(z)z-dB* (z)=Q (z-1)β*(z)-zL*(z)。
23. the approximating method of compensating filter according to claim 16, it is characterised in that: use minimum phase method by institute
It states target sound signal to be fitted with the analoging sound signal, with obtain being connected with the input terminal of the simulation system
The parameter of compensating filter method particularly includes: assuming that the transmission function of simulation system isBypass punishment filter
Transmission function are as follows:It is the β that spectral factorization equation is asked, makes multinomial of the β zero point in unit circle, spectral factorization equation
For；
β(z-1) β * (z)=V (z-1)V*(z)B(z-1)B*(z)+W(z-1)W*(z)A(z-1)A*(z)
β(z-1) β * (z)=V (z-1)V*(z)B(z-1)B*(z)+W(z-1)W*(z)A(z-1)A*(z)
By above formula, the parameter of the compensating filter can be obtained.
24. the approximating method of compensating filter according to claim 16, it is characterised in that: control simulation using LQG
The target sound signal is fitted with the analoging sound signal, to obtain being connected with the input terminal of the simulation system
The parameter of the compensating filter connect method particularly includes: on the basis of minimum phase method, it is assumed that the target sound signal is first
Time delay module d is crossed, assume that the expression formula of the compensating filter is in this wayBypass punishment filter
Transmission function are as follows:It can be in the hope of Q (q by solving Diophantine equation as follows-1) to obtain the compensation filter
The parameter of device:
z-dV(z-1) V* (z) B* (z)=Q (z-1)β*(z)-zL*(z)
z-dV(z-1) V* (z) B* (z)=Q (z-1)β*(z)-zL*(z)。
25. a kind of Sound Imitation System, which is characterized in that the Sound Imitation System includes at least:
Simulation system, including the second transmission function module, suitable for the target sound signal that provides goal systems by described the
The analoging sound signal of simulated target voice signal is exported after the second transmission function processing in two transmission function modules；
Compensating filter, including input terminal and output end；The input of the compensating filter is connected with the goal systems, institute
The output end for stating compensating filter is connected with the input terminal of the simulation system, suitable for the sound for inputting the simulation system
Signal is handled, so that the variance of the energy of the analoging sound signal and the target sound signal is minimum；The benefit
The parameter for repaying filter is obtained as the compensating filter fitting system as described in any one of claims 1 to 15.
26. Sound Imitation System according to claim 25, it is characterised in that: the simulation system and the goal systems
It is earphone or speaker.
27. Sound Imitation System according to claim 26, it is characterised in that: the simulation system is Passive earphone or nothing
Source speaker, the input terminal of the compensating filter are connected with the goal systems, the input of output end and the simulation system
End is connected.
28. Sound Imitation System according to claim 26, it is characterised in that: the simulation system is active earphone or has
Source speaker, the compensating filter are located inside the simulation system.
29. a kind of sound simulation method, which is characterized in that the sound simulation method includes at least:
1) compensating filter is provided, the parameter of the compensating filter is as the compensation as described in any one of claims 1 to 15
Filter fits system obtains；
2) target sound signal the compensating filter is inputted to handle；
3) by the compensating filter, treated that voice signal is sent in simulation system is played.
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| CN108108557B (en) * | 2017-12-21 | 2020-06-02 | 北京华大九天软件有限公司 | Adaptive fitting and simulation method for nport problem based on vector matching method |
| CN110913305B (en) * | 2019-12-05 | 2021-08-24 | 广东技术师范大学 | Self-adaptive equalizer compensation method for vehicle-mounted sound equipment |
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