CN103634726B - A kind of Automatic loudspeaker equalization method - Google Patents

A kind of Automatic loudspeaker equalization method Download PDF

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CN103634726B
CN103634726B CN201310674495.1A CN201310674495A CN103634726B CN 103634726 B CN103634726 B CN 103634726B CN 201310674495 A CN201310674495 A CN 201310674495A CN 103634726 B CN103634726 B CN 103634726B
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signal
frequency
function
equalization method
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CN103634726A (en
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叶超
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Suzhou Sonavox Electronics Co Ltd
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SHANGSHENG ELECTRONIC CO Ltd SUZHOU
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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
    • H04R29/00—Monitoring arrangements; Testing arrangements
    • H04R29/001—Monitoring arrangements; Testing arrangements for loudspeakers
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2499/00—Aspects covered by H04R or H04S not otherwise provided for in their subgroups
    • H04R2499/10—General applications
    • H04R2499/13—Acoustic transducers and sound field adaptation in vehicles
    • 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
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
    • H04S7/30—Control circuits for electronic adaptation of the sound field
    • H04S7/301—Automatic calibration of stereophonic sound system, e.g. with test microphone
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
    • H04S7/30—Control circuits for electronic adaptation of the sound field
    • H04S7/307—Frequency adjustment, e.g. tone control

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  • Health & Medical Sciences (AREA)
  • General Health & Medical Sciences (AREA)
  • Otolaryngology (AREA)
  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Circuit For Audible Band Transducer (AREA)
  • Tone Control, Compression And Expansion, Limiting Amplitude (AREA)

Abstract

The invention provides a kind of speaker automatic equalization system, improve the sound playback performance of speaker system in full frequency band.The method includes：Measure the impulse response of one or more location points in room by microphone, obtain the frequency response of each position point and the low-frequency minimum of speaker system, be equalized wave filter using adaptive optimization algorithm, speaker system is compensated.For the low frequency signal less than speaker system lower frequency limit, using based on psychoacoustic missing fundamental principle, produce the higher harmonic component of fundamental frequency signal, with the original audio signal phase superposition through time delay after gain control, improve the sound play capability in full frequency band for the speaker system.

Description

A kind of Automatic loudspeaker equalization method
Technical field
The present invention relates to Audio Signal Processing technical field is and in particular to a kind of speaker automatic equalization system, purpose exists In correction room, the frequecy characteristic of listened position, improves the sound play capability of speaker system, improves tonequality.More specifically, This equalization methods comprise virtual bass boost technology, by the non-linear harmonic component producing low-frequency component, to improve low frequency The perception of composition.
Background technology
Preferably sound-reproducing system, should have more straight frequency response in full frequency band, but in process of production due to The restriction of manufacturing process, leads to speaker system can not have preferable frequency response, and there are certain distortion.Other one Aspect, due to the interaction between the impact of room mode and speaker system and room, can not be complete at listening location Whole realization is truly low voice speaking to put.Therefore, using sound field correction technique, speaker system need to be equalized so that at one or many Frequency response at individual location point close to preferable flat curve, to ensure the true playback of primary signal.
At present, existing balancing technique has Graphic equalizer and parametric equalizer, the peak value of mainly one group cascade or slope Mode filter, the mid frequency of each wave filter corresponds to octave or third-octave, by adjusting the gain of each wave filter This frequency range is controlled, thus realizing the correction to whole frequency range.This method is more directly perceived, realizes simple, easy to operate, However it is necessary that being familiar with to the sound property of each frequency range, could more accurately debug, and the cascade of each wave filter Superposition, uncontrollable situation in the amplitude being easily caused some frequencies.In the case of more practical, first pass through microphone measurement Speaker system, in the frequency response of one or more location points, then carries out equalizer design according to the curve recording, equilibrium The form of device is FIR（Finite impulse response）Or IIR（Infinite-duration impulse response）Wave filter, is filtered to input signal, makes Obtain and obtain approximately straight frequency response in each location point.But, because the frequency resolution of low-frequency range is relatively low, therefore in order to Improving low frequency resolution needs to increase the exponent number of wave filter, increased computation complexity.In addition, for small diameter loudspeaker list Unit, if using directly in a balanced way method increase the energy of low frequency signal, replay signal can be led to distort, or even can damage and raise one's voice Device system.Can be very good to solve this problem based on the virtual bass boost technology of psychoacousticss missing fundamental principle, utilize Human ear obtains the nonlinear interaction of sound, from the subjective perception improving all-bottom sound, can improve the low frequency of small diameter loudspeaker Play capability.
Content of the invention
It is an object of the invention to provide a kind of Automatic loudspeaker equalization method, to compensate speaker self-defect and room Impact, controls audio signal loudness feature, to improve the sound playback performance at each position point.
In order to achieve the above object, the present invention provides a kind of Automatic loudspeaker equalization method, in turn includes the following steps：
1）Measure the transmission function of speaker system electrical input signal multiple location points in room using microphone；
2）Determine the prototype function of multiple transmission functions according to the weighted connections of multiple location points；
3）The equalization filter of prototype function；
4）Determined the low-frequency minimum frequency of speaker system by prototype function, and low pass filtered is carried out to original input signal Ripple, obtains the low frequency fundamental frequency signal less than lower frequency limit；
5）Produce the higher hamonic wave signal of low frequency fundamental frequency signal using nonlinear algorithm；
6）Higher hamonic wave signal after dynamic range control, with through time delay original input signal superposition rear feed give Jun Heng Wave filter；
7）Equalization filter output signal drives loudspeaker unit after power amplifier.
Further, step 1）Described in measure the method for speaker system transmission function can be using swept-frequency signal or Greatly enhance degree series（MLS）, or other method for impulse response measurement.Selected measurement position point should preferably be received in room Listen position, or cover preferred listening area, each seat position of such as home theater or automotive interior.
Further, step 2）Described in determine that the prototype function calculating process of multiple transmission functions is as follows.
AssumeIndividual location point recordsIndividual transmission function,, prototype function is to characterizeThe characterisitic function of individual transmission function common trend, can be calculated by following two modes.
A) utilizeThe weighted root mean square of individual transmission function is as prototype function
(1)
Wherein,For weight coefficient, according to practical situation, diverse location point can be weighted, such as in home theater In, more emphasize the tonequality of the position just to screen, and secondary consideration other positions.And for example in automotive interior, can basis Actual demand carries out different weights to front-seat or back row seat.When,, prototype function isIndividual transmission The root-mean-square value of function.
B) utilizeThe weighted arithmetic average of individual transmission function is as prototype function
(2)
Wherein,For weight coefficient, according to practical situation, diverse location point can be weighted.When,, Prototype function isThe arithmetic equal value of individual transmission function.
Prototype function describes the common trait of multiple location point transmission functions, and in room sound field, prototype function is from straight Reach the common denominator that the aspects such as sound, reflection and reverberation sound are extracted each location point, by the equilibrium to prototype function The sound field correction to multiple location points can be realized.
Further, step 3）The method for designing of described prototype function equalization filter can be optimum using time-domain adaptive Change algorithm, including Minimum Mean Square Error method, least square method of recursion etc..Adaptive algorithm adjusts the filtering of itself by automatic Iterative Device parameter, to meet the requirement of minimum criteria, thus realize the filter coefficient of optimum.
Further, step 4）Described speakers low frequencies lower frequency limit is determined by its physical characteristic；Low pass filter FIR can be adopted（Finite impulse response）Or IIR（Infinite-duration impulse response）Filter form.The amplitude-frequency characteristic essence of iir filter Degree is higher, and system function can be write as the form of closing function, is realized using recursion type structure, and computation complexity is relatively low, but Phase characteristic is not linear, and needs to consider system stability.And FIR filter amplitude-frequency characteristic precision will compared to IIR Low, typically no analytical expression, computation complexity is higher, and its remarkable advantage is that system is stable, and has linear The feature of phase place.
Further, step 5）The described nonlinear algorithm producing higher hamonic wave signal can be polynomial function, index Function or power function and other nonlinear functions, to produce the higher harmonic component of input low frequency signal.
Further, step 6）Described dynamic range control refers to carry out dynamic control to higher hamonic wave signal, passes through Peakvalue's checking to higher hamonic wave signal and gain control, realize the control of appreciable low frequency signal.
Further, step 7）Described power amplifier can have analog- and digital- two kinds of implementations.If adopted Simulated implementation mode, the digital signal of equalization filter output becomes analogue signal through digital-to-analogue conversion, then is entered by power amplifier Row signal power is amplified；If adopting digital implementation, the digital signal of equalization filter output is directly fed to digital power Amplifier carries out signal power amplification.
Further, step 7）Described loudspeaker unit can be the moving-coil speaker of various different sizes and specification.
Compared with prior art, it is an advantage of the current invention that：
A. the present invention, can be by speaker by on-line measurement in use environment for the speaker system and real time equaliser System is combined with room acoustical characteristic, lifts performance in specifically used environment for the speaker system.
B. the present invention is respectively processed to the low frequency of speaker system frequency response and high frequency, for less than lower frequency limit Low frequency signal carry out virtual bass boost, adaptive equalization carries out for the signal higher than lower frequency limit, improves speaker system System is in the sound play capability of full frequency band.
C. the present invention can equalize to location points multiple in room, by extracting multiple location point transmission functions Prototype function realizes multi-spot balancing, it is to avoid the tonequality after a point equilibrium in room other positions point being likely to result in is damaged Evil.
D. the present invention carries out the calculating of equalization filter using time-domain adaptive algorithm, can effectively improve essence in a balanced way Degree, and adopt time domain equalization algorithm, frequency domain algorithm can be avoided simultaneously need to consider the equilibrium of amplitude and phase place, decrease calculating Complexity.
Brief description
Fig. 1 is the signal processing flow figure of the speaker automatic equalization system of the present invention；
Fig. 2 is the flow chart realizing sound field balancing procedure in Fig. 1；
Fig. 3 is to utilize adaptive algorithm to calculate the schematic diagram of equalization filter in Fig. 2；
Fig. 4 is the signal processing flow figure realizing virtual bass boost in Fig. 1；
Fig. 5 is the signal processing flow figure realizing dynamic range control in Fig. 1；
Fig. 6 is the time-domain curve figure of the equalization filter of one embodiment of the invention；
Fig. 7 A is the speaker system time-domain pulse response curve chart of one embodiment of the invention；
Fig. 7 B be one embodiment of the invention speaker system through equilibrium after time-domain pulse response curve chart；
Fig. 8 A is the speaker system frequency response curve of one embodiment of the invention；
Fig. 8 B be one embodiment of the invention speaker system through equilibrium after frequency response curve；
Specific embodiment
With reference to the accompanying drawings and detailed description the present invention is described in further detail：
The present invention first passes through the impulse response that microphone measures speaker system multiple location points in room, by multiple Impulse response determines its prototype function, calculates the equalization filter of prototype function using adaptive optimization algorithm；According to prototype Function determines the low-frequency minimum frequency of speaker system, carries out virtual bass boost to the input signal less than this frequency, with reality Existing speaker system is in the automatic equalization of full frequency band.
The speaker automatic equalization system of the foundation present invention as shown in Figure 1, its main body is increased by sound source 101, virtual bass Strong module 102, dynamic range control 103, delay cell 104, equalization filter 105, power amplifier 106 and loudspeaker unit 107 grade compositions.Sound source 101 is connected with the input of described virtual bass boost 102, for less than speaker system lower limit The bass of frequency is strengthened；The outfan of virtual bass boost module 102 is connected with the input of dynamic range control 103, The signal processing through virtual bass is carried out with dynamic control, removes noise；The outfan of dynamic range control 103 is single with delay The outfan of unit 104 is added, then is connected with the input of equalization filter 105, carries out equilibrium treatment to input signal, then delivers to Power amplifier 106, is amplified to signal after equalization, and drive the speaker unit 107 sounding.
In Fig. 1 shown in calculating process Fig. 2 of equalization filter 105, concrete implementation step is to survey first with microphone In amount room, the impulse response of multiple location points, obtains prototype function 202 using aforesaid (1) formula or (2) formula, due to prototype letter Number 202 is generally non-minimum phase system, therefore can be classified as minimum phase system 203 and all-pass system 204, respectively To amplitude information 205 and phase information 206；Then frequency transformation 207 is carried out to it, by prototype function by linear using (3) formula Frequency transformation to inflection frequency, to improve low frequency resolution.
(3)
In (3) formula,For frequency domain delay unit,For bending the factor, span is.In bending frequency Rate domain, using adaptive optimization algorithm 208, is equalized wave filter 209.
Adaptive optimization algorithm principle figure is as shown in figure 3, whereinFor input signal,For wave filter, For system function,WithIt is respectively desired signal and output signal,For error signal.By Optimized algorithm updates wave filterCoefficient, make error signalMinimum.Preferably, taking LMSE method as a example, Illustrate the calculating process of adaptive optimization algorithm.Assume input signal vector, Filter coefficient,For filter length.Wink using single sample square-error Gradient vector is estimated in duration, that is,
(4)
The computing formula of filter coefficient is
(5)
Wherein,For step factor.Value is bigger, and algorithmic statement is faster, but steady-state error is bigger；It is worth less, algorithm Convergence is slower, but steady-state error is less.
Virtual bass boost module is as shown in Figure 4.Input signal 401 first passes around low pass filter 402, by non-linear Produce higher hamonic wave 403, after then carrying out bandpass filtering, 404 are stacked with the original input signal 401 through delay unit 405 Plus, obtain the output signal 406 through virtual bass boost.Non-linear produce harmonic wave 403 mode can be polynomial function, Exponential function or power function and other nonlinear functions.Preferably, the non-linear mode producing harmonic wave can be as (6) formula Polynomial form.
(6)
Wherein,For constant coefficient.
Dynamic range control handling process is as shown in Figure 5.Through virtual bass boost output signal 406 as dynamic model Contain the input signal 501 of system, with being originally inputted through delay unit 504 after peakvalue's checking 502 with gain control 503 Signal is multiplied, to realize the dynamic range control to original input signal 501.
The present invention will be described in detail for below in conjunction with the accompanying drawings with one embodiment.
In the present embodiment, a size of 3.5 inches of loudspeaker unit, record the arteries and veins of speaker system first with microphone Punching response is with frequency response respectively as shown in Fig. 7 A and Fig. 8 A.It is equalized wave filter using adaptive optimization method, adopt 300 rank FIR forms, time domain waveform is as shown in Figure 6.Impulse response after speaker system is equalized and frequency response are respectively As shown in Fig. 7 B and Fig. 8 B.It can be seen that after equilibrium treatment, the impulse response of speaker system is more sharp, Frequency response is more flat, has preferable frequency characteristic.And carry out bass compensation using virtual bass boost algorithms.Pass through Actual audition, low-frequency range representability is remarkably reinforced, and the music of medium-high frequency is brighter, and sound is more natural.
It should be noted last that, above example is only in order to illustrate technical scheme and unrestricted.Although ginseng According to embodiment, the present invention is described in detail, it will be understood by those within the art that, the technical side to the present invention Case is modified or equivalent, and without departure from the spirit and scope of technical solution of the present invention, it all should be covered in the present invention Right in the middle of.

Claims (11)

1. a kind of Automatic loudspeaker equalization method, in turn includes the following steps：
1) microphone is utilized to measure the transmission function of speaker system electrical input signal multiple location points in room；
2) prototype function of multiple transmission functions is determined according to the weighted connections of multiple location points；
3) equalization filter of prototype function；
4) determined the low-frequency minimum frequency of speaker system by prototype function, and low pass filtered is carried out to original electrical input signal Ripple, obtains the low frequency fundamental frequency signal less than lower frequency limit；
5) nonlinear algorithm is utilized to produce the higher hamonic wave signal of low frequency fundamental frequency signal；
6) higher hamonic wave signal is after dynamic range control, with the original electrical input signal superposition rear feed through time delay give Jun Heng Wave filter；
7) equalization filter output digit signals drive loudspeaker unit after power amplifier.
2. Automatic loudspeaker equalization method according to claim 1 it is characterised in that：Step 1) in selected measurement position Putting is a little the listened position choosing in room, or for covering the listening area of selected listened position.
3. Automatic loudspeaker equalization method according to claim 1 it is characterised in that：Step 2) Central Plains type function describes The common trait of the transmission function of one or more location points, in room sound field, prototype function from direct sound wave, reflection and The aspects such as reverberation sound are extracted the common denominator of each location point it is assumed that recording M transfer function H in M location pointi(ejω), i =1,2 ..., M, the calculation of prototype function includes,
A) by the use of M transmission function weighted root mean square as prototype function
| H ^ ( e j ω ) | = 1 M Σ i = 1 M | a i H i ( e j ω ) | 2
Wherein, aiFor weight coefficient, i=1,2 ..., M；
B) by the use of M transmission function weighted arithmetic average as prototype function
| H ^ ( e j ω ) | = 1 M Σ i = 1 M | b i H i ( e j ω ) |
Wherein, biFor weight coefficient, i=1,2 ..., M.
4. the Automatic loudspeaker equalization method according to claim 1 it is characterised in that：Step 3) in equalization filter Method for designing adopts adaptive optimization mode, including Minimum Mean Square Error method, least square method of recursion.
5. the Automatic loudspeaker equalization method according to claim 1 it is characterised in that：Step 4) under speakers low frequencies Limit frequency is determined by its physical characteristic.
6. the Automatic loudspeaker equalization method according to claim 1 it is characterised in that：Step 4) adopt a low-pass filtering Device, this low pass filter is finite impulse response or infinite impulse response filter form.
7. the Automatic loudspeaker equalization method according to claim 1 it is characterised in that：Step 5) middle generation higher hamonic wave The nonlinear algorithm of signal includes polynomial function, exponential function or power function, other nonlinear functions, to produce low frequency base The higher harmonic component of frequency signal.
8. the Automatic loudspeaker equalization method according to claim 1 it is characterised in that：Step 6) in dynamic range control Refer to carry out dynamic control to higher hamonic wave signal, by the peakvalue's checking of higher hamonic wave signal and gain control, realization can The control of the low frequency signal of perception.
9. the Automatic loudspeaker equalization method according to claim 1 it is characterised in that：Step 7) intermediate power amplifier is Simulated implementation mode, the digital signal of equalization filter output becomes analogue signal through digital-to-analogue conversion, then is entered by power amplifier Row signal power exports after amplifying, and is used for drive the speaker unit.
10. the Automatic loudspeaker equalization method according to claim 1 it is characterised in that：Step 7) intermediate power amplifier For digital implementation, the power amplifier that the digital signal of equalization filter output is directly fed to digital form carries out signal work( Rate exports after amplifying, for drive the speaker unit.
11. Automatic loudspeaker equalization method according to claim 1 it is characterised in that：Step 7) in loudspeaker unit Moving-coil speaker including multiple different sizes and specification.
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