CN102523542A - Quasi adaptive acoustic equalization method and apparatus thereof - Google Patents
Quasi adaptive acoustic equalization method and apparatus thereof Download PDFInfo
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- CN102523542A CN102523542A CN2011104172319A CN201110417231A CN102523542A CN 102523542 A CN102523542 A CN 102523542A CN 2011104172319 A CN2011104172319 A CN 2011104172319A CN 201110417231 A CN201110417231 A CN 201110417231A CN 102523542 A CN102523542 A CN 102523542A
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Abstract
A quasi adaptive acoustic equalization method is characterized by: using an adaptive digital filter and a corresponding DSP algorithm module to realize changing of a sound source signal and carry out adaptive processing to a whole audio frequency pass band; using a quasi adaptive acoustic overall equalizer, which combines modal equalization, phase equalization and amplitude equalization, to process the whole audio frequency pass band; firstly carrying out low band modal equalization to a low pass channel of a frequency division network; then superposing high and low pass sound signals in an area which is close to a frequency division point on a high pass channel and carrying out the phase equalization; and then carrying out the amplitude equalization to a whole frequency band and a high frequency band so as to realize comprehensive optimization processing to the whole audio frequency pass band. The general equalization means that a measurement microphone and several DSP algorithm modules form a modal equalizer module, a phase equalizer module and an amplitude equalizer module respectively so as to realize and have effect on a loudspeaker system. The three modules can be realized by using a digital filter with an adjustable parameter. Finally, the needed audio signal can be obtained.
Description
Technical field
The present invention relates to adaptive acoustic equalization methods and balancer,, can be widely used in the speaker system in the room according to the design of dividing network, adaptive digital filter and the realization of DSP algoritic module.
Background technology
After deliberation very thorough of theoretical side of tradition dividing network; But in the actual acoustic surrounding except that anechoic chamber; For example in the normal room; Adopt the practical manifestation of speaker system of traditional dividing network but often very big with the theoretical prediction deviation, this is owing to produced interaction between actual room and the speaker system, and this interaction; Perhaps be called the room effect and usually can make actual hearing effect variation, more common acoustical reproduction is arbitrarily downgraded, and fluctuation appears in frequency response curve, the low frequency modal reverberation time is long and the three major types problems such as peak valley of part-frequency point frequency near zone.To this three major types problem that the room effect causes, people have proposed some solutions:
A.R.Groh; " High-Fidelity Sound System Equalization by Analysis of Standing Waves, " [J] J.Audio Eng.Soc., vol.22; Pp.795-799 (1974 Dec.) can be under the situation that LisPos is fixed; Putting position through the adjustment speaker system makes its antinode of shifting to some mode standing wave or node, adjusts the acoustic pressure size that LisPos receives.
K.D.Farnsworth; P.A.Nelson; And S.J.Elliott, " Equalization of Room Acoustic Response over Spatially Distributed Regions, " [C] Proc.Inst.Acoust. (1985) is under all fixing situation of LisPos and sound source position; Through at adaptive digital filter of dividing network end cascade, realize the straight optimization aim of acoustical signal amplitude-frequency response that receives on single-point or the multiple spot LisPos
J.Mourjopoulos; " Digital Equalization Methods for Audio Systems; " [C] presented at the 84th Convention of the Audio Engineering Society, J.Audio Eng.Soc., vol.36; P.384 (1988 May) proposes the notion of " active " sound absorption; Be the Applied Digital filtering theory, through method at the extra additional a kind of mode equalizer (mode equalizer) of input electrical signal passage, can be to the resonance frequency that receives the rate of decay or reverberation time of very low room mode control; Reach the purpose of low frequency and medium-high frequency reverberation time balance, and proved the reliability of this method with experiment.
S.J.Elliot; P.A.Nelson, " Multiple-Point Equalization in a Room Using Adaptive Digital Filters, " [J] J.Audio Eng.Soc.; Vol.37; Pp.899-907 (1989Nov.) can change high low pass two interchannel phase differences, thereby improve the grazing of the sound pressure level frequency response curve of speaker system in minute frequency domain through the method in a series of second order all-pass networks of dividing network high pass channel cascade or extra introducing time-delay.
Said method has just solved the problem of (room acoustics or electroacoustics) in a certain respect, and the frequency range of improvement is also limited to (frequency range or full frequency band near low-frequency range, the part-frequency point) very much, and comprehensive inadequately, effect also is barely satisfactory, lacks a unified effective method.
Summary of the invention
The objective of the invention is: propose a kind of accurate adaptive acoustic equalization methods and device; On existing research basis about dividing network, the active sound-absorption method in the room acoustics and the phase equalization method in the electroacoustics and amplitude equalization method are combined, the new improvement method to existing dividing network is proposed; Automatically adjust the sound pressure level frequency response curve of speaker system in actual room; Make it to become straight at the full range band, and the reverberation time at control low frequency modal place, thereby the accuracy of speaker system performance in the raising theoretical prediction room; Subjective definition and the smoothness that improves signal makes it balance, nature more.
The objective of the invention is to realize like this: a kind of accurate adaptive acoustic equalization methods; The variation to sound-source signal of using adaptive digital filter and corresponding DSP algoritic module to realize is handled whole audio frequency passband adaptively, and the overall equalizer of accurate adaptive acoustic that mode equilibrium, phase equalization and amplitude equalization combine is handled whole audio frequency passband; Adopt adaptive digital filter and DSP algoritic module; Earlier the low circulation passage of dividing network is carried out the equilibrium of low-frequency range mode; In high pass channel phase equalization is carried out in the high low pass acoustical signal stack on the part-frequency point near zone then; Subsequently to whole frequency, be mainly high band and carry out amplitude equalization, thereby realize comprehensive optimization process to whole audio frequency passband; Said low circulation passage refers generally to the frequency channel less than 200Hz, and high pass channel refers to the frequency channel greater than 1000Hz; Overall equilibrium constitutes mode equalizer H respectively by measuring microphone and some DSP algoritic modules
c(z), phase equalizer H
p(z) and amplitude equalizer H
m(z) module realizes and acts on speaker system, and three module is all realized with the adjustable digital filter of parameter;
Adopt adaptive digital filter and DSP algoritic module; Earlier the low circulation passage of dividing network is carried out low-frequency range mode balanced (carrying out active sound-absorption); In high pass channel phase equalization is carried out in the high low pass acoustical signal stack on the part-frequency point near zone then; Subsequently whole frequency (being mainly high band) is carried out amplitude equalization, thereby realize comprehensive optimization process whole audio frequency passband.
Beneficial effect of the present invention is: adjust the sound pressure level frequency response curve of speaker system in actual room automatically through overall equilibrium; Active sound-absorption method in the room acoustics and the phase equalization method in the electroacoustics and amplitude equalization method are combined; The reverberation time at control low frequency modal place; Thereby the accuracy of speaker system performance in the raising room, the definition and the smoothness of raising signal make it balance, nature more.Can be optimized processing to the whole audio frequency passband of speaker system adaptively along with the variation of programme signal, universality is strong, in listening room, automotive interior etc. comprises the bounded space of speaker system, all can use, and with low cost, effect is remarkable.
Description of drawings
Fig. 1 is the overall equalizing system design of the accurate self adaptation block diagram that the present invention adopts.
Fig. 2 is according to the present invention, at the low circulation passage append mode equalizer of dividing network, and high pass channel additive phase equalizer, after the whole additional amplitude equalizer, the sound pressure level frequency response curve of 2.1 channel loudspeaker systems in actual room.
The practical implementation method
Concrete implementation method of the present invention is described below.
This overall equalizer is made up of a measurement microphone and some DSP algoritic modules, and its nucleus module is mode equalizer H
c(z), phase equalizer H
p(z) and amplitude equalizer H
m(z), the three all realizes with the adjustable digital filter of parameter.Three equalizers act on speaker system.
When reality is used,, make its transfer function be all 1 at first with three equalizer initialization.Three travel permit spare selector switches all are in closed condition.Be placed on the LisPos place that wants balanced in the room with measuring microphone, speaker system produces a random signal or swept-frequency signal x [n], through amplitude equalizer H
m(z) after, feed-in simultaneously contains mode equalizer H
c(z) dividing network hangs down circulation passage and contains phase equalizer H
p(z) high pass channel, and the corresponding DA that is changes.High low pass two-way analog signal after the conversion is sent into speaker system, and the acoustical signal that measuring microphone receives is divided into three the tunnel, is respectively applied for the parameter of confirming three kinds of equalizers.
First via acoustical signal is earlier through LPF; Get the low-frequency component below its 200Hz; Can calculate the impulse response of the total system of speaker system and room formation after the AD conversion, calculate, need the parameters (for example model frequency, mode reverberation time etc.) of balanced room mode in this low-frequency range; And calculate the desired value of reverberation time after the mode equilibrium simultaneously, it is deposited in the register.The difference of desired value in reverberation time and the register of actual measurement mode as error signal, is sent in the condition selector switch 1.The size that act as judgement error originated from input signal of this condition selector switch if error is bigger, surpasses certain limit value, then error signal is sent into mode equalizer H
c(z), help its adjustment parameter, to obtain new error signal.Size up to error signal reaches certain below the limit value, just thinks to reach the balanced target of mode, and condition selector switch 1 just cuts out itself and mode equalizer H
c(z) passage between makes mode equalizer H
c(z) parameter immobilizes, and activation condition selector switch 2.
The second tunnel acoustical signal is earlier through a band pass filter, and passband is chosen as near the frequency domain the part-frequency point frequency.After the AD conversion, can calculate the amplitude fluctuations variance of acoustical signal, and compare, form error signal with the variance desired value in the register in this frequency range.This error signal is also being controlled condition selector switch 2, has only when error signal during greater than certain limit value, and condition selector switch 2 just can be sent to phase equalizer H with it
p(z), help its change parameter; When the error signal size is lower than limit value, think that the target of phase equalization is reached, condition selector switch 2 cuts out itself and phase equalizer H
p(z) passage between makes H
p(z) parameter immobilizes, and activation condition selector switch 3.
For realizing amplitude equalization, uses a filter D (z) that presets parameter, make input electrical signal x [n] through behind this filter, produce the response of the speaker system of wanting to reach, correspondence straight sound pressure level frequency response curve.The Third Road acoustical signal is through a low pass filter that prevents aliased distortion, and after the AD conversion, direct and x [n] compares through the filtered ideal signal of D (z), and the error signal of generation is sent into condition selector switch 3.During greater than limit value, condition selector switch 3 is sent it into amplitude equalizer H at error amount
m(z), help its change parameter; After error amount was less than limit value, condition selector switch 3 cut out itself and amplitude equalizer H
m(z) passage between makes H
m(z) parameter immobilizes, and closes the signal generation passage of x [n].
So far, required mode equalizer H
c(z), phase equalizer H
p(z) and amplitude equalizer H
m(z) parameter is all being adjusted to optimum value automatically, and the user can be changed to x [n] and want the audio signal of listening.
Why cry accurate self adaptation overall equalizer; Because this system needs to handle a period of time with steady-state noise or swept-frequency signal earlier in use; After treating that parametric equalizer is fixing; Just can import the common multimedia audio signal and carry out audition, this is because the common multimedia audio signal is unknown at difference frequency spectrum constantly, can't calculate the parameter of equalizer immediately.
Claims (3)
1. accurate adaptive acoustic equalization methods; It is characterized in that the variation to sound-source signal of using adaptive digital filter and corresponding DSP algoritic module to realize handles whole audio frequency passband adaptively, the overall equalizer of accurate adaptive acoustic that mode is balanced, phase equalization and amplitude equalization combine is handled whole audio frequency passband; Adopt adaptive digital filter and DSP algoritic module; Earlier the low circulation passage of dividing network is carried out the equilibrium of low-frequency range mode; In high pass channel phase equalization is carried out in the high low pass acoustical signal stack on the part-frequency point near zone then; Subsequently to whole frequency, be mainly high band and carry out amplitude equalization, thereby realize comprehensive optimization process to whole audio frequency passband; Said low circulation passage refers generally to the frequency channel less than 200Hz, and high pass channel refers to the frequency channel greater than 1000Hz; Overall equilibrium constitutes mode equalizer H respectively by measuring microphone and some DSP algoritic modules
c(z), phase equalizer H
p(z) and amplitude equalizer H
m(z) module realizes and acts on speaker system, and three module is all realized with the adjustable digital filter of parameter.
2. accurate according to claim 1 adaptive acoustic equalization methods is characterized in that including acoustic equalizer device a kind of in mode equilibrium, phase equalization and three functions of amplitude equalization, two or three.
3. accurate according to claim 1 adaptive acoustic equalization methods is characterized in that when reality is used, and at first with three equalizer initialization, makes its transfer function be all 1; The selector switch of three modules all is in closed condition; Be placed on the LisPos place that wants balanced in the room with measuring microphone, speaker system produces a random signal or swept-frequency signal x [n], through amplitude equalizer H
m(z) after, feed-in simultaneously contains mode equalizer H
c(z) dividing network hangs down circulation passage and contains phase equalizer H
p(z) high pass channel, and the corresponding DA that is changes; High low pass two-way analog signal after the conversion is sent into speaker system, and the acoustical signal that measuring microphone receives is divided into three the tunnel, is respectively applied for the parameter of confirming three kinds of equalizers.
First via acoustical signal is earlier through LPF; Get the following low-frequency component of its 200Hz; Can calculate the impulse response of the total system of speaker system and room formation after the AD conversion; Calculate in this low-frequency range and carry out the mode equilibrium; What need balanced room mode comprises model frequency, the parameters of mode reverberation time; And calculate the desired value of reverberation time after the mode equilibrium simultaneously, it is deposited in the register that is provided with;
The difference of desired value in reverberation time of actual measurement room mode and the register as error signal, is sent in the condition selector switch 1; The size that act as judgement error originated from input signal of this condition selector switch if error is bigger, surpasses certain limit value, then error signal is sent into mode equalizer H
c(z), help mode equalizer H
c(z) adjustment parameter is to obtain new error signal; Size up to error signal reaches certain below the limit value, just thinks to reach the balanced target of mode, and condition selector switch 1 just cuts out itself and mode equalizer H
c(z) passage between makes mode equalizer H
c(z) parameter immobilizes, and activation condition selector switch 2;
Switch 1,2,3 corresponding respectively open mode equalizer H
c(z), phase equalizer H
p(z) and amplitude equalizer H
m(z) switch of module;
The second tunnel acoustical signal is earlier through a band pass filter, and passband is chosen as near the frequency domain the part-frequency point frequency; After the AD conversion, calculate the amplitude fluctuations variance of acoustical signal, and compare, form error signal with the variance desired value in the register in this frequency range; When error signal during greater than certain limit value, condition selector switch 2 just can be sent to phase equalizer H with it
p(z), help its change parameter; When the error signal size is lower than limit value, think that the target of phase equalization is reached, condition selector switch 2 cuts out itself and phase equalizer H
p(z) passage between makes H
p(z) parameter immobilizes, and activation condition selector switch 3;
For realizing amplitude equalization, uses a filter D (z) that presets parameter, make input electrical signal x [n] through behind this filter, correspondence straight sound pressure level frequency response curve; The Third Road acoustical signal is through a low pass filter that prevents aliased distortion, and after the AD conversion, direct and x [n] compares through the filtered ideal signal of D (z), and the error signal of generation is sent into condition selector switch 3; During greater than limit value, condition selector switch 3 is sent it into amplitude equalizer H at error amount
m(z), help its change parameter; After error amount was less than limit value, condition selector switch 3 cut out itself and amplitude equalizer H
m(z) passage between makes H
m(z) parameter immobilizes, and closes the signal generation passage of x [n].
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| CN107566955A (en) * | 2017-09-28 | 2018-01-09 | 广州国光音频科技有限公司 | A kind of K sings audio-visual digital reverberation system |
| CN107896359A (en) * | 2017-11-15 | 2018-04-10 | 上海昶音通讯科技有限公司 | A kind of Mobile phone horn loudness method for fast measuring based on sound pressure meter |
| CN107948862A (en) * | 2017-12-26 | 2018-04-20 | 广州励丰文化科技股份有限公司 | The automatic compensation time-delay method and system of a kind of sound-box device |
| CN112511966A (en) * | 2020-11-06 | 2021-03-16 | 华南理工大学 | Self-adaptive active frequency division method for vehicle-mounted stereo playback |
| CN110111804B (en) * | 2018-02-01 | 2021-03-19 | 南京大学 | Self-adaptive dereverberation method based on RLS algorithm |
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| CN103209368A (en) * | 2013-03-27 | 2013-07-17 | 广东欧珀移动通信有限公司 | Method and device for adjusting audio in blue-ray player |
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| CN112511966A (en) * | 2020-11-06 | 2021-03-16 | 华南理工大学 | Self-adaptive active frequency division method for vehicle-mounted stereo playback |
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| CN116437268A (en) * | 2023-06-14 | 2023-07-14 | 武汉海微科技有限公司 | Adaptive frequency division surround sound upmixing method, device, equipment and storage medium |
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Application publication date: 20120627 |