CN102883243A - Method and device for balancing frequency response of sound reproduction system through online iteration - Google Patents

Method and device for balancing frequency response of sound reproduction system through online iteration Download PDF

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CN102883243A
CN102883243A CN2012103882565A CN201210388256A CN102883243A CN 102883243 A CN102883243 A CN 102883243A CN 2012103882565 A CN2012103882565 A CN 2012103882565A CN 201210388256 A CN201210388256 A CN 201210388256A CN 102883243 A CN102883243 A CN 102883243A
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sound
equalizer
signal
feedback signal
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CN102883243B (en
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马登永
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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
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/02—Circuits for transducers, loudspeakers or microphones for preventing acoustic reaction, i.e. acoustic oscillatory feedback

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Abstract

The invention discloses a method and device for balancing frequency response of a sound reproduction system through online iteration. The method comprises the following steps of: (1) turning the system to a balancing mode, setting a sound source as a noise signal, controlling the system to play the noise signal, and acquiring and recoding a feedback signal through a sensor; (2) combining the noise signal and the feedback signal, estimating parameters of all levels of balancers through a least mean squares (LMS) criterion of step-by-step iteration; (3) cascading the all levels of the balancers to form a synthesized balancer; and (4) placing the synthesized balancer to a signal processing channel, finishing the operation of balancing the system frequency response, and turning the system to a normal play mode. The device comprises the sound source, a digital signal processor, a power amplifier, a speaker, and a feedback signal receiving module. By increasing the quantity of the cascading balancers, response balancing capability of the balancers is obviously improved, and meanwhile, by online automatic balancing, balancing operation flow is simplified, and application scenarios are wide and flexible.

Description

Sound-reproducing system frequency response equalization methods and device in line interation
Technical field
The present invention relates to a kind of frequency response equalization methods and device of sound-reproducing system, particularly a kind of sound-reproducing system frequency response equalization methods and device in line interation.
Background technology
In recent years, along with developing rapidly of large scale integrated circuit and Digital Signal Processing, also receive gradually the concern of domestic and international many research institutions and enterprise based on the sound-reproducing system response equalization problem of Digital Signal Processing, and have several companies to release several moneys with the acoustic product of response equalization function.Dirac company under the University of Uppsala (Uppsala University) has released Dirac HD Sound technology, be used for solving the frequency response curve peak valley point equilibrium of loudspeaker unit under the free field environment, also release simultaneously Dirac Live technology, be used for solving the frequency response curve peak valley point equilibrium of sound-reproducing system in the room.Hong Kong is just so summoned Co., Ltd and has been released the CONEQ technology, be used for solving the frequency response fluctuating equilibrium of speaker system in the room, this technology utilizes single microphone to gather the impulse response data of speaker system according to snakelike wiring path pointwise in loudspeaker unit the place ahead, then by specific response data Processing Algorithm, carry out the frequency response of speaker system balanced.Denmark forest-road husband (LYNGDORF) company has released the room balancing technique---Room Perfect, this technology utilizes a plurality of location points of single microphone in the room to gather the response data of speaker system, and utilizes these multiple spot response messages to finish the response curve equilibrium of hearer position.U.S. KRK company has also released sound-reproducing system response balanced product---Ergo(Enhanced Room Geometry Optimization in the room), this product also is to utilize single microphone to gather response data all around in the hearer position, and process the parameter that these response datas obtain equalizer, finish the response of hearer's location point balanced.
Above-mentioned many moneys sound-reproducing system balanced product, be used for the interior loudspeaker unit of balanced free field or the speaker system in the balanced room, the Least square estimation duration of loudspeaker unit is shorter in free field, pulse stretching is less, the design of its equalizer is comparatively simple, but the Least square estimation duration of loudspeaker unit is longer in the room, pulse stretching is very serious, the design of its equalizer is comparatively complicated, needs the response data of a plurality of locus points of associating to carry out integrated treatment.Equalization algorithm that these balanced product adopt, all be by the time-domain response sequence that gathers is analyzed in essence, it is decomposed into minimum phase response part and all-pass response part, directly the zero limit of its response is inverted the zero limit that just can obtain its inverse filter response for minimum phase response part, thereby obtains the parameter of its inverse filter response; For all-pass response part, if its phase response is steady state value in the band region of expectation, do not change with Frequency generated, the all-pass response that so this phase place is constant does not partly significantly affect the tone color of replay signal, can ignore, if but the phase response of all-pass response part can change with Frequency generated in desired frequency band, so this phase response responds the tone color generation significant change that part can cause replay signal with the all-pass of frequency change, needs to consider carry out equilibrium treatment to it.
Research Literature for speaker system equalization methods in free field and the reverberation field is more, and the representational achievement in research of some of them is as follows:
Document 1---Stephen T. Neely and Jont B. Allen, " Invertibility of a room impulse response; " J. Acoust. Soc. Am, Vol. 66, No. 1, pp. 165-169, July 1979.---the computational methods of minimum phase response part in the room response have been proposed, and to the zero limit inversion of minimum phase response, obtained the liftering response of minimum phase response, inverse filter parameter based on the minimum phase part is carried out equilibrium treatment to room response, and has confirmed by experiment algorithm complexity.
Document 2---Yoichi Haneda, Shoji Makino, and Yutaka Kaneda, " Common acoustical pole and zero modeling of room transfer functions; " IEEE Transaction on Speech and Audio Processing, Vol. 2, No. 2, pp. 320-328, April 1994.---for many groups transfer function of a plurality of location points collections in the room, propose the room transfer function blending algorithm based on common acoustics pole model, reduced the estimated parameter of room model, improved computational speed.
Document 3---Aki Harma, Matti Karjalainen, Lauri Savioja, Vesa Valimaki, Unto K. Laine, and Jyri Huopaniemi, " Frequency-warped signal processing for audio application; " J. Audio Eng. Soc., Vol. 48, and No. 11, pp. 1011-1029, November 2000.---and propose the response equalization methods based on the inflection frequency territory, (Linear Predictive Coding---LPC) method is calculated the parametric equalizer in the inflection frequency territory, thereby improves the ability of equalization to low frequency region to utilize linear predictive coding.
Traditional sound-reproducing system equalization methods, all be based on the system impulse response function is analyzed, the zero pole model of these impulse response functions of match is inverted the inverse filter response of the system that finds out again by zero limit, thereby has obtained the parametric equalizer of sound-reproducing system.The parameter estimation procedure of these methods all is to depend on least mean-square error (Least Mean Squares---LMS) single of algorithm or linear predictive coding (LPC) algorithm estimates to come the parameter of computation balance device, still exist deviation to a certain degree between this inverse filter parameter that obtains based on the single method of estimation and the desirable inverse filter parameter, these deviations will cause the sound-reproducing system frequency response curve after the equilibrium to still have comparatively significantly peak valley Characteristic fluctuation in some frequency bands, not reach yet comparatively desirable frequency response falt characteristic.Peak valley fluctuation characteristics of frequency response curves were that parameter error by equalizer causes after these were balanced, for the peak valley volt that weakens balanced rear frequency response curve is levied, need further to improve the Parameter Estimation Precision of equalizer, therefore need to seek more accurate effectively parametric equalizer method of estimation.
For having now based on single LMS or single LPC parameter estimation algorithm, existing certain error defective aspect the system equalizer parameter Estimation, need to consider to adopt repeatedly the method for iterative estimate, make estimated cascade equalizer response progressively approach the response of ideal system inverse filter by iterative operation repeatedly, reduce the parameter estimating error of equalizer, thereby guarantee that balanced rear frequency response curve has better falt characteristic.
Summary of the invention
The object of the invention is to overcome based on single LMS or single LPC parameter estimation algorithm in existing certain error defective aspect the system equalizer parameter Estimation, a kind of sound-reproducing system frequency response equalization methods and device in line interation is provided.
In order to achieve the above object, the technical solution used in the present invention is as follows: a kind of sound-reproducing system frequency response equalization methods in line interation as shown in Figure 1, comprises the steps:
(1) system is revolved put balanced mode, the setting sound source is noise signal, and then control system is play this noise signal, simultaneously transducer collection and record feedback signal;
(2) in conjunction with noise signal and feedback signal, utilize the LMS criterion of iteration step by step to estimate successively the parameter of each level equaliser;
(3) each level equaliser of cascade forms synthetic equalizer;
(4) will synthesize equalizer and place signal processing channel, then completion system frequency response equalization operation is revolved system and is put normal play mode.
Further, noise signal described in the step (1), can be white noise sequence or maximal-length sequence (Maximum Length Sequence---the noise signal that MLS) produces of nominated bandwidth, sort signal presents smooth power spectrum characteristic, to be used for training in free field or the reverberation field loud speaker to the transfer function of microphone position point.
Further, transducer collection described in the step (1) and record feedback signal, as shown in Figure 2, the type of this transducer is to determine according to the equalization target of sound-reproducing system; If sound-reproducing system only needs equalizing signal to process and power amplifier drives the frequency response fluctuation characteristic of two forming circuit systems of part institute, this transducer can be for cable or analog to digital converter the processing of circuit Least square estimation signal partly to obtain to be produced through the noise excitation; If sound-reproducing system needs the response fluctuation characteristic after equalizing circuit part and the speaker portion coupling, then this transducer is the microphone that places certain desired locations point of space, be used for record by the noise excitation produce from the loud speaker to the microphone between corresponding to the Least square estimation of Underwater Acoustic Propagation; If sound-reproducing system needs the response fluctuation characteristic after equalizing circuit part, speaker portion and three parts of external environment condition part are coupled together, then this transducer is the microphone that places a certain location point in space, this microphone need to change locus of living in, record successively a plurality of locus points upper by the noise excitation produce from the loud speaker to the microphone between corresponding to the Least square estimation of Underwater Acoustic Propagation.
Further, utilize the LMS criterion of iteration step by step to estimate successively the parameter of each level equaliser described in the step (2), this parametric equalizer estimation procedures at different levels based on the LMS criterion of iteration step by step are as follows:
A. the time domain sequences vector of supposing the input noise source signal is:
Figure DEST_PATH_IMAGE002A
?，
Wherein,
Figure DEST_PATH_IMAGE004A
It is the sampling number of noise source signal time domain discrete sequence; Suppose the time-domain pulse response sequence expression formula of sound-reproducing system:
Figure DEST_PATH_IMAGE006A
，
Wherein,
Figure DEST_PATH_IMAGE008A
Sequence length for system's time-domain pulse response; The time domain sequences of supposing the feedback signal of transducer collection is:
Figure DEST_PATH_IMAGE010A
，
Wherein "
Figure DEST_PATH_IMAGE012AA
" represent between two sequence of vectors and carry out convolution operation; As shown in Figure 3, when the 1st iteration, suppose that the length of the time-domain pulse response sequence of the 1st equalizer to be asked is
Figure DEST_PATH_IMAGE014A
, this response sequence vector can be expressed as so:
Figure DEST_PATH_IMAGE016A
，
Feedback signal Via the 1st equalizer to be asked After the processing, the feedback signal expression formula that obtains after the 1st equilibrium is:
Figure DEST_PATH_IMAGE022A
，
In conjunction with sound-source signal
Figure DEST_PATH_IMAGE024AAAAAA
With the feedback signal after the 1st equilibrium
Figure DEST_PATH_IMAGE026AA
, according to minimum mean square error criterion, calculate the sound-source signal of sening as an envoy to
Figure DEST_PATH_IMAGE024AAAAAAA
With feedback signal after the equilibrium
Figure DEST_PATH_IMAGE026AAA
Between the parametric equalizer vector of mean square error when getting minimum value, the estimated value of the 1st the optimal equaliser parameter vector that designs during as the 1st iteration, its expression formula is:
Figure DEST_PATH_IMAGE029A
，
As shown in Figure 4, can calculate the estimated value of optimal equaliser parameter vector according to the LMS algorithm, the computational process of its optimal equaliser parameter is to realize by the iterative algorithm based on LMS,
Figure DEST_PATH_IMAGE031AAAA
In the inferior iterative process, the calculation expression of the 1st optimal equaliser parameter is as follows:
Figure DEST_PATH_IMAGE033A
，
Wherein Represent sound-source signal
Figure DEST_PATH_IMAGE037A
Retardation; According to above-mentioned expression formula as can be known, the parameter iteration process of the 1st optimal equaliser as shown in Figure 5, wherein Be step-length, its value satisfies following relation:
Figure DEST_PATH_IMAGE041AA
，
Wherein
Figure DEST_PATH_IMAGE043A
Be matrix Mark (diagonal element sum),
Figure DEST_PATH_IMAGE047A
,
Figure DEST_PATH_IMAGE049A
It is vector
Figure DEST_PATH_IMAGE018AAAAA
The conjugate transpose vector, and
Figure DEST_PATH_IMAGE018AAAAAA
To be by the length that the feedback signal that transducer receives becomes
Figure DEST_PATH_IMAGE053AA
The time domain sequences vector, Individual sampling instant feedback signal
Figure DEST_PATH_IMAGE057A
Vector expression be:
Figure DEST_PATH_IMAGE059A
In addition, step-length
Figure DEST_PATH_IMAGE039AAAAAAAAAAA
Also can choose according to following relation:
Figure DEST_PATH_IMAGE061AA
，
Wherein
Figure DEST_PATH_IMAGE063AA
Be matrix
Figure DEST_PATH_IMAGE045AAAA
Eigenvalue of maximum; Step factor
Figure DEST_PATH_IMAGE039AAAAAAAAAAAA
Actual value size to select in conjunction with the practical operation situation of LMS algorithm, to guarantee that algorithm has preferably convergence rate under the less prerequisite of offset error.
In order to improve LMS convergence of algorithm speed, the calculating of optimal equaliser parameter also can be adopted normalized LMS algorithm, i.e. NLMS (Normalized Least Mean Squares) algorithm is then
Figure DEST_PATH_IMAGE031AAAAA
In the inferior iterative process, as follows based on the calculation expression of the 1st optimal equaliser parameter of NLMS algorithm:
，
Wherein
Figure DEST_PATH_IMAGE039AAAAAAAAAAAAA
Be step factor,
Figure DEST_PATH_IMAGE067AAAA
Be little normal number,
Figure DEST_PATH_IMAGE069A
For NLMS algorithm, parameter With
Figure DEST_PATH_IMAGE067AAAAA
Value also want the practical operation situation of combination algorithm to select, to guarantee in the less situation of offset error, improving convergence of algorithm speed as far as possible.
Suppose by LMS algorithm or NLMS algorithm warp
Figure DEST_PATH_IMAGE071AA
Algorithm is tending towards convergence after the inferior iterative computation, and this moment, the estimates of parameters of the 1st equalizer was:
Figure DEST_PATH_IMAGE073A
, to the estimates of parameters of the 1st equalizer
Figure DEST_PATH_IMAGE075AA
Carry out normalized, obtain:
Figure DEST_PATH_IMAGE077A
，
Wherein,
Figure DEST_PATH_IMAGE079A
It is vector
Figure DEST_PATH_IMAGE075AAA
The transposition vector; Feedback signal
Figure DEST_PATH_IMAGE018AAAAAAA
Via the 1st equalizer
Figure DEST_PATH_IMAGE082AA
After the processing, the feedback signal expression formula that obtains after the 1st equilibrium is:
Figure DEST_PATH_IMAGE084A
，
In conjunction with the feedback signal after the 1st equilibrium
Figure DEST_PATH_IMAGE086A
And sound-source signal , we can calculate feedback signal after the 1st level equaliser is processed and the root-mean-square error value between the sound-source signal , its expression formula is as follows:
Figure DEST_PATH_IMAGE091A
，
B. according to the estimation procedure of parametric equalizer among the step a, continue to finish the 1st, 2 ...,
Figure DEST_PATH_IMAGE093AAAAAA
The parameter Estimation of individual equalizer is supposed finish
Figure DEST_PATH_IMAGE093AAAAAAA
On the basis that individual parametric equalizer is estimated, as shown in Figure 6, the
Figure DEST_PATH_IMAGE095AAAAAAAAAAAAAAAAAAA
The estimation procedure of individual parametric equalizer is as follows:
Before the warp
Figure DEST_PATH_IMAGE093AAAAAAAA
Feedback signal after individual equalizer filtering is processed Can be expressed as:
Figure DEST_PATH_IMAGE099A
，
Before the warp
Figure DEST_PATH_IMAGE093AAAAAAAAA
Feedback signal after individual equalizer filtering is processed
Figure DEST_PATH_IMAGE097AAAAA
Again via to be asked
Figure DEST_PATH_IMAGE095AAAAAAAAAAAAAAAAAAAA
Individual equalizer After the processing, obtain through front
Figure DEST_PATH_IMAGE095AAAAAAAAAAAAAAAAAAAAA
Feedback signal expression formula after the individual equalizer processes is:
Figure DEST_PATH_IMAGE106A
，
In conjunction with sound-source signal
Figure DEST_PATH_IMAGE024AAAAAAAAA
And warp
Figure DEST_PATH_IMAGE095AAAAAAAAAAAAAAAAAAAAAA
Feedback signal after the individual equalizer processes , according to minimum mean square error criterion, calculate the sound-source signal of sening as an envoy to
Figure DEST_PATH_IMAGE024AAAAAAAAAA
With feedback signal after the equilibrium
Figure DEST_PATH_IMAGE110AAAA
Between the parametric equalizer vector of mean square error when getting minimum value, as
Figure DEST_PATH_IMAGE095AAAAAAAAAAAAAAAAAAAAAAA
Design during inferior iteration
Figure DEST_PATH_IMAGE095AAAAAAAAAAAAAAAAAAAAAAAA
The estimated value of individual optimal equaliser parameter vector, its expression formula is:
Figure DEST_PATH_IMAGE114A
，
As shown in Figure 6, can calculate the estimated value of optimal equaliser parameter vector according to the LMS algorithm, the computational process of its optimal equaliser parameter is to realize by the iterative algorithm based on LMS,
Figure DEST_PATH_IMAGE031AAAAAA
In the inferior iterative process, the
Figure DEST_PATH_IMAGE095AAAAAAAAAAAAAAAAAAAAAAAAA
The calculation expression of individual optimal equaliser parameter is as follows:
Figure DEST_PATH_IMAGE116A
，
According to above-mentioned expression formula as can be known,
Figure DEST_PATH_IMAGE095AAAAAAAAAAAAAAAAAAAAAAAAAA
The iterative process of individual parametric equalizer as shown in Figure 7, step-length wherein
Figure DEST_PATH_IMAGE039AAAAAAAAAAAAAAA
Value satisfy following relation:
，
Figure DEST_PATH_IMAGE118A
, It is vector
Figure DEST_PATH_IMAGE097AAAAAA
The conjugate transpose vector, and
Figure DEST_PATH_IMAGE097AAAAAAA
Through front
Figure DEST_PATH_IMAGE093AAAAAAAAAA
The length that feedback signal after individual equalizer filtering is processed forms is The time domain sequences vector,
Figure DEST_PATH_IMAGE055AAA
Before the individual sampling instant warp
Figure DEST_PATH_IMAGE093AAAAAAAAAAA
Feedback signal after individual optimal equaliser filtering is processed, the vector expression of its time domain is:
Figure DEST_PATH_IMAGE126A
In addition, step-length
Figure DEST_PATH_IMAGE039AAAAAAAAAAAAAAAA
Also can choose according to following relation:
Figure DEST_PATH_IMAGE061AAA
，
Wherein
Figure DEST_PATH_IMAGE063AAA
Be matrix
Figure DEST_PATH_IMAGE045AAAAA
Eigenvalue of maximum.Step factor
Figure DEST_PATH_IMAGE039AAAAAAAAAAAAAAAAA
Actual value size to select in conjunction with the practical operation situation of LMS algorithm, to guarantee that algorithm has preferably convergence rate under the less prerequisite of offset error.
In order to improve LMS convergence of algorithm speed, the calculating of optimal equaliser parameter also can be adopted normalized LMS algorithm, i.e. NLMS algorithm is
Figure DEST_PATH_IMAGE031AAAAAAA
In the inferior iterative process, based on of NLMS algorithm
Figure DEST_PATH_IMAGE095AAAAAAAAAAAAAAAAAAAAAAAAAAA
The calculation expression of individual optimal equaliser parameter is as follows:
Figure DEST_PATH_IMAGE129A
，
Wherein
Figure DEST_PATH_IMAGE039AAAAAAAAAAAAAAAAAA
Be step factor, Be little normal number, For NLMS algorithm, parameter
Figure DEST_PATH_IMAGE039AAAAAAAAAAAAAAAAAAA
With
Figure DEST_PATH_IMAGE067AAAAAAA
Value also want the practical operation situation of combination algorithm to select, to guarantee in the less situation of offset error, improving convergence of algorithm speed as far as possible.
Suppose by LMS algorithm or NLMS algorithm warp
Figure DEST_PATH_IMAGE071AAA
Algorithm is tending towards convergence after the inferior iterative computation, this moment the
Figure DEST_PATH_IMAGE095AAAAAAAAAAAAAAAAAAAAAAAAAAAA
The estimates of parameters of individual equalizer is: , to The estimates of parameters of individual equalizer Carry out normalized, obtain:
，
Wherein, It is vector
Figure DEST_PATH_IMAGE140AAA
The transposition vector.Before the warp
Figure DEST_PATH_IMAGE147A
Feedback signal after individual equalizer filtering is processed
Figure DEST_PATH_IMAGE149A
Again via
Figure DEST_PATH_IMAGE095AAAAAAAAAAAAAAAAAAAAAAAAAAAAAA
Individual equalizer
Figure DEST_PATH_IMAGE151A
After the processing, obtain through front Feedback signal after the individual equalizer processes, its expression formula is:
Figure DEST_PATH_IMAGE153A
，
Before warp
Figure DEST_PATH_IMAGE095AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA
Feedback signal after the individual equalizer processes
Figure DEST_PATH_IMAGE110AAAAA
And sound-source signal
Figure DEST_PATH_IMAGE024AAAAAAAAAAA
, we can calculate through front
Figure DEST_PATH_IMAGE095AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA
Feedback signal after level equaliser is processed and the root-mean-square error value between the sound-source signal
Figure DEST_PATH_IMAGE157A
, its expression formula is as follows:
Figure DEST_PATH_IMAGE159A
；
C. according to the estimation procedure of parametric equalizer among the step b, continue to finish
Figure DEST_PATH_IMAGE161AAA
,
Figure DEST_PATH_IMAGE163A
...,
Figure DEST_PATH_IMAGE165AAAAA
The parameter Estimation of individual equalizer finishes
Figure DEST_PATH_IMAGE165AAAAAA
After the parameter Estimation of individual equalizer, balanced root error amount
Figure DEST_PATH_IMAGE167AA
Expected mean square root error less than user's setting
Figure DEST_PATH_IMAGE169AA
, algorithm no longer continues to calculate the parameter of next stage equalizer, stops iteration.
Further, in the iterative process of estimation balancing device parameter, algorithm can monitor balanced root error amount always
Figure DEST_PATH_IMAGE167AAA
Size, the expected mean square root error that after each iteration is finished, all can set with the user
Figure DEST_PATH_IMAGE169AAA
Compare, and the operation of control iterative cycles.As,
Figure DEST_PATH_IMAGE161AAAA
Before inferior iteration begins, if , continue so to carry out Inferior iteration; If
Figure DEST_PATH_IMAGE175A
, then stop iteration.
Further, each level equaliser of cascade described in the step (3) forms synthetic equalizer, and its implementation procedure is as follows:
By
Figure DEST_PATH_IMAGE165AAAAAAA
This that inferior iterative estimate obtains
Figure DEST_PATH_IMAGE165AAAAAAAA
The estimates of parameters of individual equalizer is respectively:
Figure DEST_PATH_IMAGE082AAA
,
Figure DEST_PATH_IMAGE178A
...,
Figure DEST_PATH_IMAGE180A
, by this
Figure DEST_PATH_IMAGE165AAAAAAAAA
The formed synthetic equalizer of individual equalizer cascade
Figure DEST_PATH_IMAGE182A
Expression formula be:
Figure DEST_PATH_IMAGE184A
，
Wherein "
Figure DEST_PATH_IMAGE012AAA
" represent the convolution operation between the time domain vector sequence.
Further, will synthesize equalizer described in the step (4) and place signal processing channel, completion system frequency response equalization operation, it is implemented as follows:
Obtaining synthetic equalizer
Figure DEST_PATH_IMAGE187AA
The basis on, (Finite Impulse Response---FIR) filter is realized equalization operation, as shown in Figure 8 to utilize finite impulse response (FIR).The time domain sequences vector of supposing the input sound-source signal is:
Figure DEST_PATH_IMAGE189A
，
Through synthetic equalizer Sound-source signal after the processing can be expressed as:
Figure DEST_PATH_IMAGE192A
。
Sound-source signal after synthetic equalizer processes, by delivering to the loud speaker end after the power amplifier amplification, thereby drive the loud speaker radiative acoustic wave, by this equilibrium treatment operation, sound field transfer function from loud speaker to hearer's location point has obtained equilibrium, the peak valley of its transfer function response curve has obtained inhibition, thereby has improved the quality of low voice speaking discharge signal.
Further, signal processing channel described in the step (4) refers to amplitude adjustment and the filtering operation of settling signal in digital signal processor, to obtain to be fit to the signal transmission of rear class output bandwidth requirement.
Another technical scheme provided by the present invention is: a kind of sound-reproducing system frequency response equalizing device in line interation, and as shown in Figure 1, it comprises:
Sound source is system's acoustic intelligence to be reset;
Digital signal processor is connected with the output of described sound source, is used for calculating repeatedly the estimates of parameters of each level equaliser of iteration, and associating multi-stage equalizing device forms synthetic equalizer, then will synthesize equalizer and join in the signal processing channel;
Power amplifier is connected with the output of described digital signal processor, is used for the signal after the equilibrium treatment is carried out power amplification, to drive the loud speaker sounding;
Loud speaker 4 is connected with the output of described power amplifier, is used for the electroacoustic conversion so that with the sound-source signal air of resetting;
The feedback signal receiver module is connected with the output of described loud speaker, is used for collection and record reproducing to airborne acoustical signal.
Further, sound source is to come from the simulated sound source signal that various analogue means produce, the digitally encoded signal that perhaps produces for various digital devices, perhaps be the wireless network transmissions signal, the wireless network transmissions signal is the broadcast singal that sends of wireless launcher and receives the sound-source signal that obtains user's appointment with demodulation by wireless receiver.When sound source is the simulated sound source signal, need to by analog to digital converter, be the digital pattern of the input of system's appointment with analog signal conversion; When sound source is digitally encoded signal, need to this digitally encoded signal be converted in digital signal processor inside the digital pattern of the input of system's appointment; When sound source is the wireless network transmissions signal, the signal of wireless receiver demodulation need to be converted to the digital pattern of the input that system formulates.
Further, sound-reproducing system has balanced operation pattern and normal play mode, and sound source need to arrange selection according to two kinds of sound-reproducing system different mode of operations.When sound-reproducing system places the balanced operation pattern, sound source is white noise sequence or maximal-length sequence (Maximum Length Sequence---the noise signal that MLS) produces of nominated bandwidth, sort signal presents smooth power spectrum characteristic, to be used for training in free field or the reverberation field loud speaker to the transfer function of microphone position point; When sound-reproducing system was spun on the normal play mode of operation, sound source was the sound-source signal that the needs of user's appointment are reset.
Further, sound-reproducing system has balanced operation pattern and normal play mode, and digital signal processor need to be dealt with the work accordingly according to two kinds of different mode of operations that sound-reproducing system places.When sound-reproducing system places the balanced operation pattern, digital signal processor at first carries out noise signal channel signal to be processed, amplitude adjustment and filtering operation with settling signal, obtain to be fit to transmitting of rear class output bandwidth requirement, and then deliver to the power amplifier end, after receiving feedback signal, can carry out analyzing and processing to noise source signal and feedback signal, according to iteration LMS criterion, calculate the estimates of parameters of each level equaliser, then all equalizers of cascade, obtain the estimates of parameters of synthetic equalizer, upgrade at last the corresponding coefficient value of FIR filter with the estimates of parameters of synthetic equalizer, join in the signal processing channel thereby will synthesize equalizer; When sound-reproducing system places the normal play mode of operation, digital signal processor will carry out equilibrium treatment to the input signal of user's appointment according to the estimates of parameters of synthetic equalizer, and signal after the equilibrium is delivered to the power amplifier input.
Further, the input interface of power amplifier can be divided into two types, i.e. digital input interface and analog input interface.If power amplifier has digital input interface, the digital signal that then can directly send here digital signal processor carries out delivering to the loud speaker end after power amplification is processed again, so power amplifier directly is connected with digital signal processor; If power amplifier only has the analog input interface, then need to rely on digital to analog converter, the digital signal that digital signal processor is sent here is converted to carries out power amplification after the analog signal again and processes, deliver at last the loud speaker end, so be connected with digital to analog converter between power amplifier and the digital signal processor.
Further, loud speaker is not limited to single loudspeaker unit, the way of realization of loud speaker can be single loudspeaker unit, also can be the loudspeaker array of a plurality of loudspeaker units compositions, and the shape of this array can be arranged according to loudspeaker unit quantity and practical application request, forms the various array shapes that are suitable for practical application request.
Further, the feedback signal receiver module, it is realized and working method need to be determined according to the equalization target of sound-reproducing system.If sound-reproducing system only needs equalizing signal processing and power amplifier to drive the frequency response fluctuation characteristic of two forming circuit systems of part institute, the feedback signal receiver module then receives and gathers the signal of power amplifier output, and the Serial No. that gathers is sent in the digital signal processor; If sound-reproducing system needs the response fluctuation characteristic after equalizing circuit part and the speaker portion coupling, then the feedback signal receiver module microphone that will be opposite to certain desired locations point of space receives signal and gathers, and the Serial No. that gathers is sent in the digital signal processor; If sound-reproducing system needs the response fluctuation characteristic after equalizing circuit part, speaker portion and three parts of external environment condition part are coupled together, then the feedback signal receiver module will gather the microphone reception signal that places successively a plurality of location points place, space, and the Serial No. corresponding to a plurality of locus points that will gather is delivered in the digital signal processor.
Compared with prior art, the invention has the advantages that:
1. with traditional comparing based on single LMS or single LPC parameter estimation algorithm, sound-reproducing system frequency response equalization methods in line interation proposed by the invention, by the quantity that increases the cascade equalizer and the exponent number that increases every level equaliser, can obviously promote the channel response compensation ability of synthetic equalizer, make the entire system frequency response curve after the equilibrium more straight, and make synthetic equalizer approach desirable inverse filter response;
2. the sound-reproducing system frequency response equalization methods in line interation proposed by the invention to the performance change of the residing environmental change of sound-reproducing system and loudspeaker unit itself, can be finished online the parameter Estimation of equalizer and upgrade processing.The user is when the space environment that the change loudspeaker unit of sound-reproducing system and install bin bulk properties or change sound-reproducing system are placed, only need sound-reproducing system revolved and put balanced mode, system can send noise signal automatically, and by microphone reception and record feedback signal, based on the analysis of noise and feedback signal, finish in real time parameter Estimation and the updating task of synthetic equalizer.This response compensation way based on on-line automatic equilibrium more can satisfy practical application request, has also simplified the equalization operation flow process simultaneously, has saved the equalization operation time, and its application is more extensive and flexible;
3. repeatedly iteration equalizing method proposed by the invention, can be by increasing the number of times of iteration equalizing, namely increase the quantity of cascade equalizer, realize more straight equilibrium treatment is carried out in frequency response in the whole broadband of user's expectation, its ability of equalization for frequency response in the low-frequency band will obviously be better than traditional equalization methods;
4. traditional parametric equalizer method of estimation, conversion that need to be by time-frequency domain realizes complicated cumbersome to obtain the minimum phase response component.Compare with traditional parametric equalizer method of estimation, repeatedly iteration equalizing method proposed by the invention, directly in time domain inner analysis noise signal and feedback signal, and directly finish the parameter Estimation of equalizer in time domain, its signal processing flow and hardware are realized comparatively simple;
5. the present invention adopts the iterative algorithm of LMS or NLMS to come the parameter value of estimation balancing device, and this parameter estimation algorithm can realize in the digital signal processors such as DSP and FPGA that fully hardware is realized simply, cost is lower;
6. the present invention generates single synthetic equalizer by a plurality of equalizers of having estimated being carried out cascade, in actual applications, carries out the equalization operation of passage with single synthetic equalizer, realizes simple and reliable.
Description of drawings
Fig. 1 represents of the present invention a kind of at the sound-reproducing system frequency response equalization methods of line interation and the signal processing flow figure of device;
Fig. 2 represent of the present invention a kind of in line interation sound-reproducing system frequency response equalization methods and the equilibrium treatment schematic diagram of three kinds of different links of device;
Fig. 3 represent of the present invention a kind of in line interation sound-reproducing system frequency response equalization methods and the parameter Estimation schematic diagram of the multi-stage equalizing device of device, wherein s is sound source, r is feedback signal,
Figure 110968DEST_PATH_IMAGE194
Be the sound-source signal after equilibrium treatment;
Fig. 4 represents the parameter Estimation schematic diagram of the present invention's the 1st grade of optimal equaliser in the 1st iterative process, wherein
Figure 399866DEST_PATH_IMAGE196
Be feedback signal, Be error signal,
Figure 372075DEST_PATH_IMAGE200
Be white noise,
Figure 874601DEST_PATH_IMAGE202
Be the sound-source signal after the liftering compensation;
Fig. 5 represents the parameter iteration schematic diagram of the present invention's the 1st grade of optimal equaliser in the 1st iterative process, wherein
Figure 599980DEST_PATH_IMAGE204
Be feedback signal,
Figure 187956DEST_PATH_IMAGE198
Be error signal,
Figure 659913DEST_PATH_IMAGE206
Be step-length,
Figure 16945DEST_PATH_IMAGE208
Be parametric equalizer;
Fig. 6 represents that the present invention is
Figure DEST_PATH_IMAGE095AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA
In the inferior iterative process
Figure DEST_PATH_IMAGE095AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA
The parameter Estimation schematic diagram of level optimal equaliser, wherein
Figure 52508DEST_PATH_IMAGE212
Be feedback signal, Be error signal,
Figure 324800DEST_PATH_IMAGE200
Be white noise,
Figure 801918DEST_PATH_IMAGE202
Be the sound-source signal after the liftering compensation;
Fig. 7 represents that the present invention is
Figure DEST_PATH_IMAGE095AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA
In the inferior iterative process
Figure DEST_PATH_IMAGE095AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA
The parameter iteration schematic diagram of level optimal equaliser, wherein
Figure 987874DEST_PATH_IMAGE212
Be feedback signal, Be error signal,
Figure 361273DEST_PATH_IMAGE206
Be step-length,
Figure 633510DEST_PATH_IMAGE217
Be parametric equalizer;
Fig. 8 represents the implementation procedure schematic diagram of synthetic equalizer of the present invention, wherein
Figure 871593DEST_PATH_IMAGE219
Be the input sound-source signal, Be the input signal after synthetic equalizer processes;
Fig. 9 represents the schematic diagram that respectively forms module of a kind of sound-reproducing system frequency response equalizing device in line interation of the present invention;
Figure 10 represents the time domain waveform figure of system works noise source signal when balanced mode in the embodiment of the invention;
Figure 11 represents the system works feedback signal waveform figure that microphone receives when balanced mode in the embodiment of the invention;
Figure 12 represent in the embodiment of the invention system do not apply equilibrium, through 1 iteration equalizing and behind 10 iteration equalizings system's frequency response curve comparison diagram.
Wherein number in the figure is:
1, source of sound; 2, digital signal processor; 3, power amplifier; 4, loud speaker; 5, feedback signal receiver module.
Embodiment
Below in conjunction with accompanying drawing preferred embodiment of the present invention is described in detail, thereby so that advantages and features of the invention can be easier to be it will be appreciated by those skilled in the art that protection scope of the present invention is made more explicit defining.
At present, traditional sound-reproducing system equalization methods all is based on the system impulse response function is analyzed, the zero pole model of these impulse response functions of match, be inverted again the inverse filter response of the system that finds out by zero limit, thereby obtained the parametric equalizer of sound-reproducing system.The parameter estimation procedure of these methods all is the parameter that the single that depends on least mean-square error (LMS) algorithm or linear predictive coding (LPC) algorithm estimates to come the computation balance device, still exist deviation to a certain degree between this inverse filter parameter that obtains based on the single method of estimation and the desirable inverse filter parameter, these deviations will cause the sound-reproducing system frequency response curve after the equilibrium to still have comparatively significantly peak valley Characteristic fluctuation in some frequency bands, not reach yet comparatively desirable frequency response falt characteristic.In order to overcome based on single LMS or single LPC parameter estimation algorithm, existing certain error defective aspect the system equalizer parameter Estimation, the present invention proposes a kind of sound-reproducing system frequency response equalization methods and device in line interation, by adopting repeatedly iterative estimate method step-by-step calculation to go out the parameter value of a plurality of cascade equalizers, the formed synthetic equalizer of these equalizer cascades at different levels can better approach the response of ideal system inverse filter, thereby reduced the parameter estimating error of equalizer, guaranteed that balanced rear system frequency response curve has better falt characteristic.The present invention can obviously promote the response ability of equalization of equalizer by increasing cascade equalizer quantity, simultaneously by on-line automatic equilibrium, has simplified the equalization operation flow process, and its application scenarios is more extensive and flexible.
As shown in Figure 9, make a foundation sound-reproducing system frequency response equalizing device in line interation of the present invention, its main body is comprised of sound source 1, digital signal processor 2, power amplifier 3, loud speaker 4, feedback signal receiver module 5 etc.
Sound source 1, sound source 1 is white noise signal when putting balanced mode when system revolves, and its sample rate is 23.8KHz, and number of bits is 16, and its time domain signal waveform is as shown in figure 10.When system revolves when putting normal play mode, sound source 1 is the replay signal for the treatment of of user's appointment.
Digital signal processor 2 is connected with the output of described sound source 1, can be realized as core processor by DSP or FPGA on hardware is realized.Under balanced mode, digital signal processor 2 will be united noise signal and feedback signal, calculate the parameter of each level equaliser by Iterative Method repeatedly, on the basis of finishing the estimation of multi-stage equalizing device, all equalizers of cascade form synthetic equalizer, and utilize the FIR filter to realize synthetic equalizer.Under normal play mode, digital signal processor 2 will utilize to be treated replay signal based on the synthetic equalizer of FIR structure and carries out equilibrium treatment.
Power amplifier 3 is connected with the output of described digital signal processor 2, and the digital signal that digital signal processor 2 is sent into is carried out digital to analog conversion and power amplification processing.
Loud speaker 4 is connected with the output of described power amplifier 3, realizes the electroacoustic conversion, acoustical reproduction signal in air.Loud speaker 4 is that 3.5 cun, rated power are that 10 watts, D.C. resistance are 4 ohm, place the loud speaker in the closed box for bore.
Feedback signal receiver module 5 is connected with the output of described loud speaker 4, and under balanced mode, the feedback signal receiver module will gather the response sequence by generation that noise source encourages, and deliver in the digital signal processor 2.
Embodiment:
In the present embodiment, suppose that sound-reproducing system works under the balanced mode, loud speaker is that sound source is white noise signal shown in Figure 10, and microphone places 1 meter on the loudspeaker unit axis, and the feedback signal time domain waveform that is recorded by microphone as shown in figure 11.The exponent number of supposing each level equaliser to be estimated is 600, and the number of times that iteration equalizing is set is 10.
Figure 12 provided balanced, through 1 iteration equalizing with in three kinds of situations of 10 iteration equalizings, the comparison diagram of system's frequency response curve.Contrasting this three suites line can find out, exists very significantly peak value not applying in the situation of equalizer system's frequency response curve in the frequency band range of 1.5KHz～4.5 KHz; After the 1st iterative equalization process, the peak value of system in the frequency band range at 1.5KHz～4.5KHz obtained elimination, but system's frequency response curve still has fluctuating in a small amount near the zone the 1.5KHz frequency, in the frequency band of 100Hz～200Hz, system's frequency response curve still has largely and rises and falls simultaneously; After 10 iterative equalization process, a small amount of fluctuating of system in 1.5KHz frequency near zone obtained elimination, and the amplitude peak in 100Hz～200Hz frequency band has also obtained suppressing largely simultaneously.Contrast the frequency response curve after the 1st iteration equalizing and the 10th iterative equalization process, can find out: by increasing iterations, iteration equalizing method proposed by the invention, can obviously improve the flatness of frequency response curve behind the system equalization, the repeatedly iteration equalizing method that this explanation is proposed by the invention, more traditional equalization methods is compared, and has better portfolio effect, and the frequency response curve after its equilibrium will be more straight.
Above-described embodiment only is explanation technical conceive of the present invention and characteristics, and its purpose is to allow the personage who is familiar with technique can understand content of the present invention and according to this enforcement, can not limit protection scope of the present invention with this.All equivalences that Spirit Essence is done according to the present invention change or modify, and all should be encompassed within protection scope of the present invention.

Claims (14)

1. the sound-reproducing system frequency response equalization methods in line interation comprises the steps:
(1) system is revolved put balanced mode, the setting sound source is noise signal, and then control system is play this noise signal, simultaneously transducer collection and record feedback signal;
(2) in conjunction with noise signal and feedback signal, utilize the minimum mean square error criterion of iteration step by step to estimate successively the parameter of each level equaliser;
(3) each level equaliser of cascade forms synthetic equalizer;
(4) will synthesize equalizer and place signal processing channel, then completion system frequency response equalization operation is revolved system and is put normal play mode.
2. the sound-reproducing system frequency response equalization methods in line interation according to claim 1 is characterized in that: the noise signal in the described step (1), the noise signal that produces for white noise sequence or the maximal-length sequence of nominated bandwidth.
3. the sound-reproducing system frequency response equalization methods in line interation according to claim 1, it is characterized in that: the utilization in the described step (2) the step by step minimum mean square error criterion of iteration estimates the parameter of each level equaliser successively, and this parametric equalizer estimation procedures at different levels based on the minimum mean square error criterion of iteration step by step are as follows:
A. the time domain sequences vector of supposing the input noise source signal is:
Figure 218857DEST_PATH_IMAGE002
，
Wherein,
Figure 245587DEST_PATH_IMAGE004
It is the sampling number of noise source signal time domain discrete sequence; The time-domain pulse response sequence expression formula of supposing sound-reproducing system is:
，
Wherein,
Figure 344921DEST_PATH_IMAGE008
Sequence length for system's time-domain pulse response; The time domain sequences of supposing the feedback signal of transducer collection is:
，
Wherein " " represent between two sequence of vectors and carry out convolution operation; When the 1st iteration, suppose that the length of the time-domain pulse response sequence of the 1st equalizer to be asked is
Figure 173965DEST_PATH_IMAGE014
, this response sequence vector then is expressed as:
Figure 658429DEST_PATH_IMAGE016
，
Feedback signal
Figure 240589DEST_PATH_IMAGE018
Via the 1st equalizer to be asked
Figure 976333DEST_PATH_IMAGE020
After the processing, the feedback signal expression formula that obtains after the 1st equilibrium is:
，
In conjunction with sound-source signal
Figure 708239DEST_PATH_IMAGE024
With the feedback signal after the 1st equilibrium
Figure 766194DEST_PATH_IMAGE026
, according to minimum mean square error criterion, calculate the sound-source signal of sening as an envoy to
Figure DEST_PATH_IMAGE027
With feedback signal after the equilibrium
Figure 2012103882565100001DEST_PATH_IMAGE028
Between the parametric equalizer vector of mean square error when getting minimum value, the estimated value of the 1st the optimal equaliser parameter vector that designs during as the 1st iteration, its expression formula is:
Figure 2012103882565100001DEST_PATH_IMAGE030
；
Calculate the estimated value of optimal equaliser parameter vector according to least-mean-square error algorithm, the computational process of its optimal equaliser parameter is to realize by the iterative algorithm based on least mean-square error,
Figure 2012103882565100001DEST_PATH_IMAGE032
In the inferior iterative process, the calculation expression of the 1st optimal equaliser parameter is as follows:
Figure 2012103882565100001DEST_PATH_IMAGE034
，
Wherein
Figure 2012103882565100001DEST_PATH_IMAGE036
Represent sound-source signal
Figure 2012103882565100001DEST_PATH_IMAGE038
Retardation; According to above-mentioned expression formula as can be known, the parameter iteration process of the 1st optimal equaliser, wherein
Figure 2012103882565100001DEST_PATH_IMAGE040
Be step-length, its value satisfies following relation:
Figure 2012103882565100001DEST_PATH_IMAGE042
，
Wherein
Figure 2012103882565100001DEST_PATH_IMAGE044
Be matrix
Figure 2012103882565100001DEST_PATH_IMAGE046
Mark,
Figure 2012103882565100001DEST_PATH_IMAGE048
,
Figure 2012103882565100001DEST_PATH_IMAGE050
It is vector
Figure DEST_PATH_IMAGE051
The conjugate transpose vector, and
Figure 132277DEST_PATH_IMAGE051
To be by the length that the feedback signal that transducer receives becomes
Figure 578171DEST_PATH_IMAGE053
The time domain sequences vector,
Figure 191555DEST_PATH_IMAGE055
Individual sampling instant feedback signal Vector expression be:
Figure 94362DEST_PATH_IMAGE059
Step-length Choose according to following relation: ,
Wherein
Figure 2012103882565100001DEST_PATH_IMAGE064
Be matrix Eigenvalue of maximum;
Suppose by the least-mean-square error algorithm warp
Figure 2012103882565100001DEST_PATH_IMAGE066
Algorithm is tending towards convergence after the inferior iterative computation, and this moment, the estimates of parameters of the 1st equalizer was:
Figure 2012103882565100001DEST_PATH_IMAGE068
, to the estimates of parameters of the 1st equalizer
Figure 2012103882565100001DEST_PATH_IMAGE070
Carry out normalized, obtain:
Figure 117562DEST_PATH_IMAGE071
，
Wherein,
Figure 848145DEST_PATH_IMAGE073
It is vector
Figure 412988DEST_PATH_IMAGE070
The transposition vector;
Feedback signal
Figure 203614DEST_PATH_IMAGE018
Via the 1st equalizer After the processing, the feedback signal expression formula that obtains after the 1st equilibrium is:
Figure DEST_PATH_IMAGE077
，
In conjunction with the feedback signal after the 1st equilibrium
Figure DEST_PATH_IMAGE079
And sound-source signal
Figure 241907DEST_PATH_IMAGE080
, calculate feedback signal after the 1st level equaliser is processed and the root-mean-square error value between the sound-source signal
Figure 2012103882565100001DEST_PATH_IMAGE082
, its expression formula is as follows:
，
B. according to the estimation procedure of parametric equalizer among the step a, continue to finish the 1st, 2 ...,
Figure 2012103882565100001DEST_PATH_IMAGE086
The parameter Estimation of individual equalizer is supposed finish
Figure 339963DEST_PATH_IMAGE086
On the basis that individual parametric equalizer is estimated, the
Figure 2012103882565100001DEST_PATH_IMAGE088
The estimation procedure of individual parametric equalizer is as follows:
Before the warp
Figure 339405DEST_PATH_IMAGE086
Feedback signal after individual equalizer filtering is processed
Figure 2012103882565100001DEST_PATH_IMAGE090
Be expressed as:
Figure DEST_PATH_IMAGE092
，
Before the warp
Figure 397536DEST_PATH_IMAGE086
Feedback signal after individual equalizer filtering is processed
Figure 738388DEST_PATH_IMAGE093
Again via to be asked Individual equalizer
Figure DEST_PATH_IMAGE095
After the processing, obtain through front
Figure 412525DEST_PATH_IMAGE088
Feedback signal expression formula after the individual equalizer processes is:
，
In conjunction with sound-source signal
Figure DEST_PATH_IMAGE098
And warp Feedback signal after the individual equalizer processes , according to minimum mean square error criterion, calculate the sound-source signal of sening as an envoy to
Figure DEST_PATH_IMAGE102
With feedback signal after the equilibrium
Figure 592675DEST_PATH_IMAGE101
Between the parametric equalizer vector of mean square error when getting minimum value, as
Figure 605104DEST_PATH_IMAGE088
Design during inferior iteration
Figure 999045DEST_PATH_IMAGE088
The estimated value of individual optimal equaliser parameter vector, its expression formula is:
,
Calculate the estimated value of optimal equaliser parameter vector according to least-mean-square error algorithm, the computational process of its optimal equaliser parameter is to realize by the iterative algorithm based on least mean-square error,
Figure DEST_PATH_IMAGE105
In the inferior iterative process, the
Figure 476873DEST_PATH_IMAGE088
The calculation expression of individual optimal equaliser parameter is as follows:
,
According to above-mentioned expression formula as can be known,
Figure 12766DEST_PATH_IMAGE099
The iterative process of individual parametric equalizer, wherein step-length
Figure DEST_PATH_IMAGE107
Value satisfy following relation:
Figure 778990DEST_PATH_IMAGE108
,
,
Figure 524803DEST_PATH_IMAGE112
It is vector
Figure DEST_PATH_IMAGE113
The conjugate transpose vector, and
Figure 917914DEST_PATH_IMAGE114
Through front The length that feedback signal after individual equalizer filtering is processed forms is
Figure 993896DEST_PATH_IMAGE053
The time domain sequences vector,
Figure 80670DEST_PATH_IMAGE055
Before the individual sampling instant warp
Figure 23612DEST_PATH_IMAGE086
Feedback signal after individual optimal equaliser filtering is processed, the vector expression of its time domain is:
Step-length
Figure 371285DEST_PATH_IMAGE060
Choose according to following relation:
Figure 125002DEST_PATH_IMAGE062
,
Wherein
Figure 558126DEST_PATH_IMAGE064
Be matrix
Figure 52562DEST_PATH_IMAGE046
Eigenvalue of maximum;
Suppose by the least-mean-square error algorithm warp Algorithm is tending towards convergence after the inferior iterative computation, this moment the
Figure 332024DEST_PATH_IMAGE088
The estimates of parameters of individual equalizer is:
Figure DEST_PATH_IMAGE118
, to
Figure 317691DEST_PATH_IMAGE088
The estimates of parameters of individual equalizer
Figure 412555DEST_PATH_IMAGE120
Carry out normalized, obtain:
Figure DEST_PATH_IMAGE121
，
Wherein,
Figure DEST_PATH_IMAGE123
It is vector
Figure 49335DEST_PATH_IMAGE124
The transposition vector; Before the warp
Figure 331805DEST_PATH_IMAGE126
Feedback signal after individual equalizer filtering is processed Again via
Figure 816238DEST_PATH_IMAGE088
Individual equalizer
Figure 74568DEST_PATH_IMAGE130
After the processing, obtain through front
Figure 400376DEST_PATH_IMAGE099
Feedback signal after the individual equalizer processes, its expression formula is:
，
Before warp
Figure 95056DEST_PATH_IMAGE133
Feedback signal after the individual equalizer processes
Figure 797302DEST_PATH_IMAGE101
And sound-source signal , calculate through front Feedback signal after level equaliser is processed and the root-mean-square error value between the sound-source signal
Figure DEST_PATH_IMAGE136
, its expression formula is as follows:
Figure DEST_PATH_IMAGE138
C. according to the estimation procedure of parametric equalizer among the step b, continue to finish ,
Figure DEST_PATH_IMAGE142
...,
Figure DEST_PATH_IMAGE144
The parameter Estimation of individual equalizer finishes
Figure 733081DEST_PATH_IMAGE145
After the parameter Estimation of individual equalizer, balanced root error amount
Figure 665789DEST_PATH_IMAGE147
Expected mean square root error less than user's setting
Figure 906147DEST_PATH_IMAGE149
, algorithm no longer continues to calculate the parameter of next stage equalizer, stops iteration.
4. the sound-reproducing system frequency response equalization methods in line interation according to claim 3, it is characterized in that: in the iterative process of estimation balancing device parameter, algorithm monitors balanced root error amount always
Figure 932877DEST_PATH_IMAGE147
Size, the expected mean square root error that after each iteration is finished, all can set with the user
Figure 665735DEST_PATH_IMAGE149
Compare, and the operation of control iterative cycles, before next iteration begins, if
Figure 473023DEST_PATH_IMAGE151
, continue so to carry out next iteration, if
Figure 305019DEST_PATH_IMAGE153
, then stop iteration.
5. the sound-reproducing system frequency response equalization methods in line interation according to claim 3, it is characterized in that: in described step a, the calculating of described optimal equaliser parameter can also be adopted normalized least-mean-square error algorithm, then
Figure 514153DEST_PATH_IMAGE032
In the inferior iterative process, as follows based on the calculation expression of the 1st optimal equaliser parameter of normalized least-mean-square error algorithm:
Figure DEST_PATH_IMAGE154
Wherein
Figure 480228DEST_PATH_IMAGE060
Be step factor,
Figure DEST_PATH_IMAGE156
Be little normal number,
Figure 636796DEST_PATH_IMAGE158
6. the sound-reproducing system frequency response equalization methods in line interation according to claim 1, it is characterized in that: in described step a, the calculating of described optimal equaliser parameter can also be adopted normalized least-mean-square error algorithm, then
Figure 156639DEST_PATH_IMAGE032
In the inferior iterative process, based on of normalized least-mean-square error algorithm The calculation expression of individual optimal equaliser parameter is as follows:
Figure DEST_PATH_IMAGE159
Wherein
Figure 419573DEST_PATH_IMAGE060
Be step factor,
Figure 888205DEST_PATH_IMAGE156
Be little normal number,
Figure DEST_PATH_IMAGE161
7. the sound-reproducing system frequency response equalization methods in line interation according to claim 1 is characterized in that: each level equaliser of cascade in the described step (3), form synthetic equalizer, and its implementation procedure is as follows:
By
Figure 336373DEST_PATH_IMAGE145
This that inferior iterative estimate obtains
Figure 319766DEST_PATH_IMAGE145
The estimates of parameters of individual equalizer is respectively:
Figure 500080DEST_PATH_IMAGE075
,
Figure DEST_PATH_IMAGE163
...,
Figure DEST_PATH_IMAGE165
, by this
Figure 880508DEST_PATH_IMAGE144
The formed synthetic equalizer of individual equalizer cascade
Figure DEST_PATH_IMAGE167
Expression formula be:
Figure DEST_PATH_IMAGE169
，
Wherein " " represent the convolution operation between the time domain vector sequence.
8. the sound-reproducing system frequency response equalization methods in line interation according to claim 1, it is characterized in that: the equalizer that will synthesize in the described step (4) places signal processing channel, completion system frequency response equalization operation, it is implemented as follows:
Obtaining synthetic equalizer The basis on, utilize finite impulse response filter to realize equalization operation, suppose the input sound-source signal the time domain sequences vector be:
Figure DEST_PATH_IMAGE173
，
Through synthetic equalizer
Figure DEST_PATH_IMAGE174
Sound-source signal after the processing can be expressed as:
Figure DEST_PATH_IMAGE176
；
Sound-source signal after synthetic equalizer processes by delivering to the loud speaker end after the power amplifier amplification, drives the loud speaker radiative acoustic wave.
9. the sound-reproducing system frequency response equalizing device in line interation according to claim 1, it is characterized in that: the signal processing channel in the described step (4), refer to amplitude adjustment and the filtering operation of settling signal in digital signal processor, obtain to be fit to the signal transmission that the rear class output bandwidth requires.
10. sound-reproducing system frequency response equalizing device in line interation, it is characterized in that: it comprises sound source (1), be connected with the output of described sound source (1) and be used for calculating repeatedly iteration each level equaliser estimates of parameters and unite the multi-stage equalizing device and form synthetic equalizer and then will synthesize the digital signal processor (2) that equalizer joins signal processing channel, be connected with the output of described digital signal processor (2) and be used for the signal after the equilibrium treatment is carried out power amplification to drive the power amplifier (3) of loud speaker sounding, described power amplifier (3), be connected with the output of described power amplifier (3) and be used for the electroacoustic conversion so that with the sound-source signal airborne loud speaker (4) of resetting, be connected with the output of described loud speaker (4) and be used for gathering and record reproducing arrives the feedback signal receiver module (5) of airborne acoustical signal, described sound source is system's acoustic intelligence to be reset.
11. the sound-reproducing system frequency response equalizing device in line interation according to claim 10, it is characterized in that: described sound source (1) is for coming from the simulated sound source signal that various analogue means produce, the digitally encoded signal that perhaps produces for various digital devices, perhaps be the wireless network transmissions signal, described wireless network transmissions signal is the broadcast singal that sends of wireless launcher and receives the sound-source signal that obtains user's appointment with demodulation by wireless receiver.
12. the sound-reproducing system frequency response equalizing device in line interation according to claim 10, it is characterized in that: described sound-reproducing system has balanced operation pattern and normal play mode of operation, when sound-reproducing system during in the balanced operation pattern, the noise signal that described sound source (1) produces for the white noise sequence of nominated bandwidth or maximal-length sequence, when sound-reproducing system was spun on the normal play mode of operation, described sound source (1) was the sound-source signal that the needs of user's appointment are reset.
13. the sound-reproducing system frequency response equalizing device in line interation according to claim 10, it is characterized in that: when described power amplifier (3) when having digital input interface, then described power amplifier (3) directly is connected with digital signal processor (2); When described power amplifier (3) only has the analog input interface, then be connected with the digital to analog converter that the digital signal that digital signal processor (2) is sent here is converted to analog signal between described power amplifier (3) and the digital signal processor (2).
14. the sound-reproducing system frequency response equalizing device in line interation according to claim 10 is characterized in that: described loud speaker (4) can be arranged loudspeaker array according to loudspeaker unit quantity and practical application request for single loudspeaker unit or for what a plurality of loudspeaker units formed.
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WO2014059890A1 (en) * 2012-10-15 2014-04-24 苏州上声电子有限公司 Method and device for balancing frequency response of sound playback system through online iteration
WO2014153943A1 (en) * 2013-03-29 2014-10-02 苏州上声电子有限公司 Balancing apparatus for sound field in vehicle
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