CN103634726A - Automatic loudspeaker equalization method - Google Patents
Automatic loudspeaker equalization method Download PDFInfo
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- CN103634726A CN103634726A CN201310674495.1A CN201310674495A CN103634726A CN 103634726 A CN103634726 A CN 103634726A CN 201310674495 A CN201310674495 A CN 201310674495A CN 103634726 A CN103634726 A CN 103634726A
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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
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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
- 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
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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
- 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
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- 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
Abstract
The invention provides an automatic loudspeaker equalization system, particularly relates to an automatic loudspeaker equalization method, and aims to improve sound replaying performance of a loudspeaker system on full bands. The method includes measuring pulse responses at one or more position points in a room by a microphone to acquire a frequency response of each position point and low-frequency lower limit of the loudspeaker system, and acquiring an equilibrium filter by the adaptive optimization algorithm to compensate the loudspeaker system. For low-frequency signals with frequency lower than the lower-limit frequency of the loudspeaker system, ultraharmonics components of fundamental-frequency signals are generated according to the psychoacoustics based fundamental-frequency missing principle and are superposed with delayed original sound signals after being subjected to gain control, and accordingly, the sound replaying performance of the loudspeaker system on full bands is improved.
Description
Technical field
The present invention relates to Audio Signal Processing technical field, be specifically related to a kind of loud speaker automatic equalization system, object is to proofread and correct the frequecy characteristic of listening to position in room, improves the sound play capability of speaker system, improves tonequality.More specifically, this equalization methods comprises virtual bass enhancing technology, by the harmonic component of non-linear generation low-frequency component, to improve the perception of low-frequency component.
Background technology
Desirable sound-reproducing system, should have more straight frequency response at full frequency band, but in process of production due to the restriction of manufacturing process, causes speaker system can not have desirable frequency response, and have certain distortion.In addition on the one hand, due to the impact of room mode and the interaction between speaker system and room, at listening location place, can completely does not realize true low voice speaking putting.Therefore, need to adopt sound field correction technique to carry out equilibrium to speaker system, make to approach desirable flat curve in the frequency response at one or more location points place, to guarantee the true playback of primary signal.
At present, existing balancing technique has Graphic equalizer and parametric equalizer, mainly peak value or the slope mode filter of one group of cascade, the centre frequency of each filter is corresponding to octave or third-octave, by adjusting the gain of each filter, this frequency range is controlled, thereby realized the correction to whole frequency range.This method is more directly perceived, realizes simply, easy to operate, but need to be familiar to some extent to the sound property of each frequency range, could debug more accurately, and the cascade superposition of each filter, easily cause the amplitude of some frequency to occur uncontrollable situation.More practical in the situation that, first by microphone, measure speaker system in the frequency response of one or more location points, then according to the curve recording, carry out equalizer design, the form of equalizer is FIR(finite impulse response) or IIR(infinite impulse response) filter, input signal is carried out to filtering, make to obtain approximate straight frequency response at each location point.But, because the frequency resolution of low-frequency range is lower, therefore in order to improve low frequency resolution, need to increase the exponent number of filter, increased computation complexity.In addition, for small diameter loudspeaker unit, if adopt directly balanced method to increase the energy of low frequency signal, replay signal distortion can be caused, even speaker system can be damaged.Virtual bass based on psychologic acoustics fundamental frequency disappearance principle strengthens technology and can well address this problem, and utilizes people's ear to obtain the nonlinear interaction of sound, can, from the subjective perception that improves all-bottom sound, improve the low frequency play capability of small diameter loudspeaker.
summary of the invention
The object of this invention is to provide a kind of loud speaker automatic balancing method, to compensate the impact in loud speaker self-defect and room, control audio signal loudness feature, to improve the sound playback performance at each location point place.
In order to achieve the above object, the invention provides a kind of loud speaker automatic balancing method, in turn include the following steps:
1) utilize microphone to measure speaker system electrical input signal to the transfer function of a plurality of location points in room;
2) according to the weighting relation of a plurality of location points, determine the prototype function of a plurality of transfer functions;
3) equalization filter of prototype function;
4) by prototype function, determined the low-frequency minimum frequency of speaker system, and original input signal is carried out to low-pass filtering, obtain the low frequency fundamental frequency signal lower than lower frequency limit;
5) utilize nonlinear algorithm to produce the high order harmonic component signal of low frequency fundamental frequency signal;
6) high order harmonic component signal is after dynamic range control, with the original input signal superposition rear feed through time delay to equalization filter;
7) equalization filter output signal is through power amplifier rear drive loudspeaker unit.
Further, the method for measuring speaker system transfer function described in step 1) can adopt swept-frequency signal or maximal-length sequence (MLS), or other method for impulse response measurement.Selected measuring position point should be in room, preferably to listen to position, or region, for example each seat position of home theater or automotive interior are preferably listened in covering.
Further, the prototype function computational process of determining a plurality of transfer functions step 2) is as follows.
Suppose
individual location point records
individual transfer function
,
, prototype function is to characterize
the characterisitic function of individual transfer function common trend, can be calculated by following two kinds of modes.
A) utilize
the weighted root mean square of individual transfer function is as prototype function
Wherein,
for weight coefficient, can to diverse location point, be weighted according to actual conditions, for example, in home theater, more emphasize over against the tonequality of the position of screen, and other position of less important consideration.And for example, in automotive interior, can to front row or back row seats, carry out different weights according to the actual requirements.When
,
, prototype function is
the root-mean-square value of individual transfer function.
B) utilize
the weighting arithmetic equal value of individual transfer function is as prototype function
Wherein,
for weight coefficient, can to diverse location point, be weighted according to actual conditions.When
,
, prototype function is
the arithmetic equal value of individual transfer function.
Prototype function has been described the common trait of a plurality of location point transfer functions, in room sound field, prototype function has been extracted the denominator of each location point from aspects such as direct sound wave, reflection and reverberation sounds, by realizing the sound field correction to a plurality of location points to the equilibrium of prototype function.
Further, the method for designing of prototype function equalization filter can adopt time-domain adaptive optimization algorithm described in step 3), comprises Minimum Mean Square Error method, least square method of recursion etc.Adaptive algorithm regulates the filter parameter of self by automatic Iterative, to meet the requirement of minimum criteria, thereby realize optimum filter coefficient.
Further, described in step 4), loud speaker low-frequency minimum frequency is determined by its physical characteristic; Low pass filter can adopt FIR(finite impulse response) or IIR(infinite impulse response) filter form.The amplitude-frequency characteristic precision of iir filter is higher, and system function can be write as the form of sealing function, adopts recursion type structure to realize, and computation complexity is lower, but phase characteristic is not linear, and needs taking into account system stability.And that FIR filter amplitude-frequency characteristic precision compares to IIR is low, generally there is no analytical expression, computation complexity is higher, and its remarkable advantage is that system is stable, and has the feature of linear phase.
Further, the nonlinear algorithm that produces high order harmonic component signal described in step 5) can be polynomial function, exponential function or power function and other nonlinear functions, to produce the high order harmonic component composition of input low frequency signal.
Further, the dynamic range control described in step 6) refers to dynamically to be controlled high order harmonic component signal, by the peak value to high order harmonic component signal, detects and gain is controlled, and realizes the control of appreciable low frequency signal.
Further, the power amplifier described in step 7) can have analog-and digital-two kinds of implementations.If adopt simulation implementation, the digital signal of equalization filter output becomes analog signal through digital-to-analogue conversion, then carries out signal power amplification by power amplifier; If adopt Digital Implementation mode, the digital signal of the equalization filter output digital power amplifier of directly feeding carries out signal power amplification.
Further, the loudspeaker unit described in step 7) can be the moving-coil speaker of various different sizes and specification.
Compared with prior art, the invention has the advantages that:
A. the present invention, by the on-line measurement in environment for use, with in real time balanced, can combine speaker system with room acoustical characteristic to speaker system, promotes the performance of speaker system in concrete environment for use.
B. the present invention processes respectively the low frequency of speaker system frequency response and high frequency, for the low frequency signal lower than lower frequency limit, carry out virtual bass enhancing, for the signal higher than lower frequency limit, carry out adaptive equalization, improve speaker system in the sound play capability of full frequency band.
C. the present invention can carry out equilibrium to a plurality of location points in room, by extracting the prototype function of a plurality of location point transfer functions, realizes multi-spot balancing, has avoided the tonequality infringement to causing other location points after some equilibrium in room.
D. the present invention adopts time-domain adaptive algorithm to carry out the calculating of equalization filter, can effectively improve balanced precision, and adopts time domain equalization algorithm, can avoid frequency domain algorithm need consider the equilibrium of amplitude and phase place simultaneously, has reduced computation complexity.
Accompanying drawing explanation
Fig. 1 is the signal processing flow figure of loud speaker automatic equalization system of the present invention;
Fig. 2 is the flow chart of realizing sound field balancing procedure in Fig. 1;
Fig. 3 utilizes adaptive algorithm to calculate the schematic diagram of equalization filter in Fig. 2;
Fig. 4 realizes the signal processing flow figure that virtual bass strengthens in Fig. 1;
Fig. 5 is the signal processing flow figure that realizes 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 is the speaker system of one embodiment of the invention time-domain pulse response curve chart after equilibrium;
Fig. 8 A is the speaker system frequency response curve of one embodiment of the invention;
Fig. 8 B is the speaker system of one embodiment of the invention frequency response curve after equilibrium;
embodiment
Below in conjunction with the drawings and specific embodiments, the present invention is described in further detail:
First the present invention measures the impulse response of speaker system a plurality of location points in room by microphone, by a plurality of impulse response, determine its prototype function, utilizes adaptive optimization algorithm to calculate the equalization filter of prototype function; According to prototype function, determine the low-frequency minimum frequency of speaker system, the input signal lower than this frequency is carried out to virtual bass enhancing, to realize speaker system in the automatic equalization of full frequency band.
As shown in Figure 1 according to loud speaker automatic equalization system of the present invention, its main body strengthens module 102, dynamic range control 103, delay cell 104, equalization filter 105, power amplifier 106 and loudspeaker unit 107 etc. by sound source 101, virtual bass and forms.Sound source 101 and described virtual bass strengthen 102 input and are connected, and for the bass to lower than speaker system lower frequency limit, strengthen; The output that virtual bass strengthens module 102 is connected with the input of dynamic range control 103, and the signal of processing through virtual bass is dynamically controlled, and removes noise; The output of the output of dynamic range control 103 and delay cell 104 is added, be connected with the input of equalization filter 105 again, input signal is carried out to equilibrium treatment, then deliver to power amplifier 106, signal after equilibrium is amplified, and drive loudspeaker unit 107 sounding.
In Fig. 1 shown in computational process Fig. 2 of equalization filter 105, concrete performing step is, first utilize microphone to measure the impulse response of a plurality of location points in room, utilize aforesaid (1) formula or (2) formula to obtain prototype function 202, because prototype function 202 is generally non minimum phase system, therefore minimum phase system 203 and all-pass system 204 be can be divided into, amplitude information 205 and phase information 206 obtained respectively; Then utilize (3) formula to carry out frequency translation 207 to it, prototype function is converted into inflection frequency by linear frequency, to improve low frequency resolution.
In (3) formula,
for frequency domain delay unit,
for the bending factor, span is
.In inflection frequency territory, utilize adaptive optimization algorithm 208, obtain equalization filter 209.
Adaptive optimization algorithm principle figure as shown in Figure 3, wherein
for input signal,
for filter,
for system function,
with
be respectively desired signal and output signal,
for error signal.By optimization algorithm, upgrade filter
coefficient, make error signal
minimum.Preferably, take least mean-square error method as example, illustrate the computational process of adaptive optimization algorithm.Suppose input signal vector
, filter coefficient
,
for filter length.Utilize the instantaneous value of single sample error square to estimate gradient vector,
The computing formula of filter coefficient is
Wherein,
for step factor.
be worth greatlyr, algorithmic statement is faster, but steady-state error is larger;
be worth littlely, algorithmic statement is slower, but steady-state error is less.
Virtual bass strengthens module as shown in Figure 4.First input signal 401 passes through low pass filter 402, by non-linear generation high order harmonic component 403, then carries out after bandpass filtering 404 superimposedly with the original input signal 401 through delay unit 405, obtains the output signal 406 strengthening through virtual bass.The mode of non-linear generation harmonic wave 403 can be polynomial function, exponential function or power function and other nonlinear functions.Preferably, the mode of non-linear generation harmonic wave can be as the polynomial form of (6) formula.
Wherein,
for constant coefficient.
Dynamic range control handling process as shown in Figure 5.The output signal 406 strengthening through virtual bass is as the input signal 501 of dynamic range control, through peak value, detect 502 and control after 503 and multiply each other with the original input signal through delay unit 504 with gain, to realize the dynamic range control to original input signal 501.
Below in conjunction with accompanying drawing and an embodiment, the present invention will be described in detail.
In the present embodiment, loudspeaker unit is of a size of 3.5 inches, first utilizes microphone to record the impulse response of speaker system and frequency response respectively as shown in Fig. 7 A and Fig. 8 A.Utilize adaptive optimization method to obtain equalization filter, adopt 300 rank FIR forms, time domain waveform as shown in Figure 6.Speaker system is carried out to impulse response after equilibrium and frequency response respectively as shown in Fig. 7 B and Fig. 8 B.As can be seen from the figure, after equilibrium treatment, the impulse response of speaker system is more sharp-pointed, and frequency response is more smooth, has good frequency characteristic.And utilize virtual bass boost algorithms to carry out bass compensation.By actual audition, low-frequency range expressive force obviously strengthens, and the music of medium-high frequency is brighter, and sound is more natural.
It should be noted last that, above embodiment is only unrestricted in order to technical scheme of the present invention to be described.Although the present invention is had been described in detail with reference to embodiment, those of ordinary skill in the art is to be understood that, technical scheme of the present invention is modified or is equal to replacement, do not depart from the spirit and scope of technical solution of the present invention, it all should be encompassed in the middle of claim scope of the present invention.
Claims (11)
1. a loud speaker automatic balancing method, in turn includes the following steps:
Utilize microphone to measure speaker system electrical input signal to the transfer function of a plurality of location points in room;
According to the weighting relation of a plurality of location points, determine the prototype function of a plurality of transfer functions;
The equalization filter of prototype function;
By prototype function, determined the low-frequency minimum frequency of speaker system, and original electrical input signal is carried out to low-pass filtering, obtain the low frequency fundamental frequency signal lower than lower frequency limit;
Utilize nonlinear algorithm to produce the high order harmonic component signal of low frequency fundamental frequency signal;
High order harmonic component signal after dynamic range control, with through the original electrical input signal superposition rear feed of time delay to equalization filter;
Equalization filter output digit signals is through power amplifier rear drive loudspeaker unit.
2. loud speaker automatic balancing method according to claim 1, is characterized in that: in step 1), selected measuring position point is to choose the position of listening in room, or for covering the selected region of listening to of listening to position.
3. loud speaker automatic balancing method according to claim 1, it is characterized in that: step 2) mesarcs function representation the common trait of transfer function of one or more location point, in room sound field, prototype function has been extracted the denominator of each location point from aspects such as direct sound wave, reflection and reverberation sounds, suppose
individual location point records
individual transfer function
,
, the account form of prototype function comprises,
Utilize
the weighted root mean square of individual transfer function is as prototype function
Wherein,
for weight coefficient,
;
Utilize
the weighting arithmetic equal value of individual transfer function is as prototype function
Wherein,
for weight coefficient,
.
4. according to the loud speaker automatic balancing method described in claim 1, it is characterized in that: in step 3), the method for designing of equalization filter adopts adaptive optimization mode, comprises Minimum Mean Square Error method, least square method of recursion.
5. according to the loud speaker automatic balancing method described in claim 1, it is characterized in that: in step 4), loud speaker low-frequency minimum frequency is determined by its physical characteristic.
6. according to the loud speaker automatic balancing method described in claim 1, it is characterized in that: step 4) adopts a low pass filter, this low pass filter is finite impulse response or infinite impulse response filter form.
7. according to the loud speaker automatic balancing method described in claim 1, it is characterized in that: the nonlinear algorithm that produces high order harmonic component signal in step 5) comprises polynomial function, exponential function or power function, other nonlinear functions, to produce the high order harmonic component composition of low frequency fundamental frequency signal.
8. according to the loud speaker automatic balancing method described in claim 1, it is characterized in that: in step 6), dynamic range control refers to high order harmonic component signal is dynamically controlled, by the peak value to high order harmonic component signal, detect and gain control, realize the control of appreciable low frequency signal.
9. according to the loud speaker automatic balancing method described in claim 1, it is characterized in that: step 7) intermediate power amplifier is simulation implementation, the digital signal of equalization filter output becomes analog signal through digital-to-analogue conversion, by power amplifier, undertaken after signal power is amplified exporting again, and for driving loudspeaker unit.
10. according to the loud speaker automatic balancing method described in claim 1, it is characterized in that: step 7) intermediate power amplifier is Digital Implementation mode, the digital signal of equalization filter output directly feed the power amplifier of digital form carry out signal power amplify after output, for driving loudspeaker unit.
11. according to the loud speaker automatic equalization system described in claim 1, it is characterized in that: in step 7), loudspeaker unit comprises the moving-coil speaker of multiple different size and specification.
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| WO2015085924A1 (en) | 2015-06-18 |
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