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 PDFInfo
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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
Technical Field
The present invention relates to a method and an apparatus for equalizing frequency response of an audio playback system, and more particularly, to a method and an apparatus for equalizing frequency response of an online iterative audio playback system.
Background
In recent years, with the rapid development of large-scale integrated circuits and digital signal processing technologies, the problem of response equalization of an audio playback system based on the digital signal processing technology is also gradually concerned by a plurality of research institutions and enterprises at home and abroad, and several companies have introduced several acoustic products with response equalization functions. The Dirac HD Sound technology is introduced by Dirac corporation under the u.p.a University (Uppsala University) flag to solve the peak-to-valley point equalization of the frequency response curve of the speaker unit in the free field environment, and the Dirac Live technology is also introduced to solve the peak-to-valley point equalization of the frequency response curve of the Sound reproduction system in the room. The CONEQ technology is provided by hong Kong just-handed telegraph Limited and is used for solving the problem of frequency response fluctuation equalization of a loudspeaker system in a room. Danish forest doff (LYNGDORF) introduced Room equalization, the Room Perfect, which uses a single microphone to collect response data from a speaker system at multiple points in a Room and uses this multi-point response information to perform response curve equalization at the listener's location. The KRK company in the united states also introduced an indoor sound reproduction system response equalization product, ergo (enhanced Room Geometry optimization), which also utilizes a single microphone to collect response data around the listener position and processes the response data to obtain parameters of an equalizer, thereby completing response equalization to the listener position point.
The equalization product of the multiple sound reproduction systems is used for equalizing the loudspeaker units in the free field or equalizing the loudspeaker systems in the room, the duration time of the impulse response sequence of the loudspeaker units in the free field is short, the impulse tailing is small, the design of the equalizer is simple, but the duration time of the impulse response sequence of the loudspeaker units in the room is long, the impulse tailing is very serious, the design of the equalizer is complex, and the comprehensive processing needs to be carried out by combining the response data of a plurality of spatial position points. The equalization algorithms adopted by these equalization products are essentially that the acquired time domain response sequence is analyzed and decomposed into a minimum phase response part and an all-pass response part, and the zero pole of the response of the minimum phase response part is directly inverted aiming at the zero pole of the response of the minimum phase response part to obtain the zero pole of the response of the inverse filter, so as to obtain the parameters of the response of the inverse filter; for the all-pass response part, if the phase response of the all-pass response part is a constant value in the expected frequency band region and does not change with the frequency, the tone color of the replay signal is not obviously influenced by the all-pass response part with the constant phase, and can be ignored, but if the phase response of the all-pass response part changes with the frequency in the expected frequency band, the all-pass response part with the phase response changing with the frequency can cause the tone color of the replay signal to obviously change, and needs to be considered for equalization processing.
There are many research documents on the equalization method of the loudspeaker system in the free field and the reverberation field, and some representative research results are as follows:
document 1-Stephen t, Neely and joint b, Allen, "inverse stability of a room impulse response," j, acout, soc, Am, vol, 66, number 1, pp. 165, 169, July 1979 "-proposes a method for calculating the minimum phase response part in the room response, and obtains the inverse filter response of the minimum phase response for the zero-inversion pole of the minimum phase response, performs equalization processing on the room response based on the inverse filter parameters of the minimum phase part, and proves the effectiveness of the algorithm through experiments.
Document 2-Yoichi Handea, Shoji Makino, and Yutaka Kaneda, "Common acoustical pole and zero modeling of room transfer functions," IEEE Transaction on Speech and Audio Processing, Vol.2, number 2, pp. 320 and 328, April 1994 "-for a plurality of sets of transfer functions collected at a plurality of location points in a room, a room transfer function fusion algorithm based on a Common acoustical pole model is proposed, which reduces the estimation parameters of the room model and improves the calculation speed.
Document 3-Aki Harma, mati Karjalainen, Lauri Savioja, Vesa Valimaki, uno k. Laine, and Jyri husbandienmi, "Frequency-weighted signal processing for Audio application," j. Audio end, soc, vol. 48, number 11, pp. 1011 1029, November 2000.— proposes a response equalization method based on a warped Frequency domain, which calculates equalizer parameters in the warped Frequency domain by using a Linear Predictive Coding (LPC) method, thereby improving the equalization capability for the low Frequency domain.
The traditional equalization method of the sound reproduction system is based on the analysis of the impulse response function of the system, the zero pole model of the impulse response function is fitted, and the inverse filter response of the system is found out through the inversion of the zero pole, so that the equalizer parameter of the sound reproduction system is obtained. The parameter estimation process of these methods relies on the single estimation of the Least Mean square error (LMS) algorithm or Linear Predictive Coding (LPC) algorithm to calculate the equalizer parameters, and there still exists a certain degree of deviation between the inverse filter parameters obtained based on the single estimation method and the ideal inverse filter parameters, and these deviations will cause the frequency response curve of the equalized sound reproduction system to have more obvious peak-valley fluctuation in some frequency bands and still not reach the more ideal frequency response flatness characteristic. The peak-to-valley fluctuation characteristics of the equalized frequency response curve are caused by the parameter error of the equalizer, and in order to weaken the peak-to-valley fluctuation characteristics of the equalized frequency response curve, the parameter estimation precision of the equalizer needs to be further improved, so that a more accurate and effective equalizer parameter estimation method needs to be found.
Aiming at certain error defects existing in the aspect of system equalizer parameter estimation based on the existing single LMS or single LPC parameter estimation algorithm, a method of multiple iterative estimation needs to be considered, the estimated cascade equalizer response gradually approaches to the ideal system inverse filter response through multiple iterative operations, the parameter estimation error of the equalizer is reduced, and therefore the frequency response curve after equalization has better straight characteristic.
Disclosure of Invention
The invention aims to overcome the defect of certain error in the aspect of system equalizer parameter estimation based on a single LMS or single LPC parameter estimation algorithm, and provides an online iterative sound reproduction system frequency response equalization method and device.
In order to achieve the purpose, the technical scheme adopted by the invention is as follows: an on-line iterative audio playback system frequency response equalization method, as shown in fig. 1, includes the following steps:
(1) setting a sound source as a noise signal in a system rotary equalization mode, then controlling the system to play the noise signal, and simultaneously collecting and recording a feedback signal by a sensor;
(2) combining the noise signal and the feedback signal, and sequentially estimating parameters of each stage of equalizer by using a step-by-step iterative LMS (least mean square) criterion;
(3) cascading the equalizers at all levels to form a synthesized equalizer;
(4) and placing the synthesized equalizer in a signal processing channel to finish the frequency response equalization operation of the system, and then placing the system in a normal play mode.
Further, the noise signal in step (1) may be a noise signal generated by a white noise Sequence or a Maximum Length Sequence (MLS) with a specified bandwidth, and such a signal exhibits a flat power spectrum characteristic for training a transfer function from a speaker to a microphone location point in a free field or a reverberation field.
Further, in step (1) the sensor collects and records a feedback signal, as shown in fig. 2, the type of such sensor is determined according to the equalization target of the sound reproduction system; if the sound reproducing system only needs to balance the frequency response fluctuation characteristics of a circuit system formed by the signal processing part and the power amplifier driving part, the sensor can be a cable or an analog-digital converter to obtain an impulse response sequence signal of the circuit processing part generated by noise excitation; if the sound reproduction system requires the response fluctuation characteristic after the equalization circuit part and the loudspeaker part are coupled, the sensor is a microphone which is arranged at a certain expected position point in the space and is used for recording an impulse response sequence which is generated by noise excitation and corresponds to the sound field propagation from the loudspeaker to the microphone; if the sound reproduction system requires response fluctuation characteristics after the three parts of the equalization circuit part, the loudspeaker part and the external environment part are coupled together, the sensor is a microphone which is arranged at a certain position point in space, the microphone needs to change the position in space, and an impulse response sequence which is generated by noise excitation at a plurality of position points in space and corresponds to sound field propagation is recorded in sequence from the loudspeaker to the microphone.
Further, in step (2), the parameters of each stage of equalizer are sequentially estimated by using the step-by-step iterative LMS criterion, and the estimation process of each stage of equalizer parameters based on the step-by-step iterative LMS criterion is as follows:
a. assume that the time domain sequence vector of the input noise source signal is:
wherein,
the number of sampling points of a time domain discrete sequence of a noise source signal; assuming the time domain impulse response sequence expression of the acoustic playback system:
wherein,
is the sequence length of the system time domain impulse response; suppose that the time domain sequence of the feedback signal collected by the sensor is:
wherein "
"represents a convolution operation between two sequence vectors; as shown in fig. 3, in the 1 st iteration, the length of the time domain impulse response sequence of the 1 st equalizer to be solved is assumed to be
Then the response sequence vector can be expressed as:
feedback signalVia the 1 st equalizer to be solvedAfter processing, the expression of the feedback signal after 1 st equalization is obtained as follows:
combined with sound source signals
And the 1 st equalized feedback signal
Calculating the sound source signal according to the minimum mean square error criterion
And the equalized feedback signal
The equalizer parameter vector when the mean square error between the two is the minimum is used as the estimated value of the 1 st optimal equalizer parameter vector designed in the 1 st iteration, and the expression is as follows:
as shown in fig. 4, the estimated value of the optimal equalizer parameter vector can be calculated according to the LMS algorithm, and the calculation process of the optimal equalizer parameter is performed by the iterative algorithm based on the LMS, in the second place
In the secondary iteration process, the calculation expression of the 1 st optimal equalizer parameter is as follows:
whereinRepresenting sound source signals
The delay amount of (3); from the above expressions, the parameter iteration process of the 1 st optimal equalizer is shown in FIG. 5, in whichThe step length is the length of the step, and the value of the step length satisfies the following relation:
wherein
Is a matrixThe trace (sum of diagonal elements),
，
is a vector
The conjugate of (2) transposes the vector, and
is a length of feedback signal received by the sensor
In the second place ofFeedback signal of sampling time
The vector expression of (a) is:
(ii) a In addition, the step size
It can also be chosen according to the following relation:
wherein
Is a matrix
The maximum eigenvalue of (d); step size factor
The actual value of the LMS algorithm is selected by combining the actual running condition of the LMS algorithm, so that the algorithm has better convergence rate on the premise of smaller misadjustment error.
In order to improve the convergence rate of the LMS algorithm, the calculation of the optimal equalizer parameters may also adopt a normalized LMS algorithm, i.e. NLMS (normalized Least Mean Square) algorithm, in the second place
In the secondary iteration process, the 1 st optimal equalizer parameter calculation expression based on the NLMS algorithm is as follows:
，
wherein
For the purpose of the step-size factor,
is a small positive constant number of the cells,
. For NLMS algorithm, parametersAnd
the value of (A) is also selected by combining the actual running condition of the algorithm so as to ensure that the convergence speed of the algorithm is improved as much as possible under the condition of small misadjustment error.
The assumption is made by LMS algorithm or NLMS algorithm
The algorithm tends to converge after the iterative computation, and the parameter estimation value of the 1 st equalizer is as follows:
estimation of parameters for the 1 st equalizer
Carrying out normalization processing to obtain:
wherein,
is a vector
Is transposed vector of(ii) a Feedback signal
Via the 1 st equalizer
After processing, the expression of the feedback signal after 1 st equalization is obtained as follows:
combining the 1 st equalized feedback signals
And sound source signalThe root mean square error value between the feedback signal processed by the 1 st stage equalizer and the sound source signal can be calculatedThe expression is as follows:
b. according to the estimation process of the equalizer parameters in the step a, continuously completing the 1 st, 2 nd, … th,
Estimation of parameters of the equalizer, assuming that
Based on equalizer parameter estimation, as shown in FIG. 6
The estimation procedure for each equalizer parameter is as follows:
before menstruation
Feedback signal after filtering processing of equalizerCan be expressed as:
before menstruation
Feedback signal after filtering processing of equalizer
Then via the first one to be asked
EqualizerAfter treatment, a pre-treatment is obtained
The expression of the feedback signal after the equalizer processing is as follows:
combined with sound source signals
And the warp
Feedback signal processed by equalizerCalculating the sound source signal according to the minimum mean square error criterion
And the equalized feedback signal
The equalizer parameter vector when the mean square error between them is minimum is taken as the second
Design at time of sub-iteration
The estimated value of the optimal equalizer parameter vector is expressed as:
as shown in fig. 6, the estimated value of the optimal equalizer parameter vector can be calculated according to the LMS algorithm, and the calculation process of the optimal equalizer parameter is performed by the iterative algorithm based on the LMS, in the second place
In the course of the second iteration, the
The computational expression for each optimal equalizer parameter is as follows:
according to the above expression, the first
The iterative process of equalizer parameters is shown in FIG. 7, where the step size is
The values of (A) satisfy the following relations:
，
wherein
Is a matrix
The maximum eigenvalue of (c). Step size factor
The actual value of the LMS algorithm is selected by combining the actual running condition of the LMS algorithm, so that the algorithm has better convergence rate on the premise of smaller misadjustment error.
To improve the convergence rate of the LMS algorithm, the calculation of the optimal equalizer parameters may also use the normalized LMS algorithm, i.e., NLMS algorithm, in the second place
In the process of secondary iteration, based on NLMS algorithm
The computational expression for each optimal equalizer parameter is as follows:
wherein
For the purpose of the step-size factor,is a small positive constant number of the cells,. For NLMS algorithm, parameters
And
the value of (A) is also selected by combining the actual running condition of the algorithm so as to ensure that the convergence speed of the algorithm is improved as much as possible under the condition of small misadjustment error.
The assumption is made by LMS algorithm or NLMS algorithm
The algorithm tends to converge after the sub-iterative computation, when the first time
The parameter estimates for each equalizer are:to the secondEstimation of equalizer parametersCarrying out normalization processing to obtain:
，
wherein,is a vector
The transposed vector of (1). Before menstruation
Feedback signal after filtering processing of equalizer
Then via the first
Equalizer
After treatment, a pre-treatment is obtainedThe feedback signal after the equalizer processing has the expression:
combined with premenstrual syndrome
Feedback signal processed by equalizer
And sound source signal
We can calculateBefore menstruation
Root mean square error value between feedback signal and sound source signal after stage equalizer processing
The expression is as follows:
c. according to the estimation process of the equalizer parameter in step b, continuously completing the second step
、
、…、
Parameter estimation of an equalizer, in the first place
After parameter estimation of the equalizer, the root error value is equalized
Less than the desired root mean square error set by the user
And the algorithm does not continue to calculate the parameters of the next-stage equalizer, and the iteration is stopped.
Further, the algorithm may monitor the equalization root error value during iterative calculations to estimate equalizer parameters
At each iteration of the size ofAfter the process is finished, the error is equal to the expected root mean square error set by a user
A comparison is made and the operation of the iterative loop is controlled. E.g. in
Before the start of the sub-iteration, ifThen continue executionPerforming secondary iteration; if it is not
The iteration is stopped.
Further, the equalizers of each stage are cascaded in step (3) to form a composite equalizer, and the implementation process is as follows:
by
This obtained by sub-iterative estimation
The parameter estimation values of the equalizers are respectively:
、
、…、
from this
Composite equalizer formed by cascading equalizers
The expression of (a) is:
wherein "
"represents a convolution operation between time domain vector sequences.
Further, the step (4) of placing the synthesized equalizer in the signal processing channel to complete the system frequency response equalization operation is specifically implemented as follows:
in obtaining a composite equalizer
Based on the above, a Finite Impulse Response (FIR) filter is used to implement the equalization operation, as shown in fig. 8. Suppose that the time domain sequence vector of the input sound source signal is:
synthesized equalizerThe processed sound source signal may be represented as:
the sound source signal processed by the synthesis equalizer is amplified by the power amplifier and then sent to the loudspeaker end, so that the loudspeaker is driven to radiate sound waves.
Further, the signal processing channel in step (4) refers to performing amplitude adjustment and filtering operations of the signal in the digital signal processor to obtain a transmission signal suitable for the requirement of the output bandwidth of the subsequent stage.
The other technical scheme provided by the invention is as follows: an on-line iterative audio playback system frequency response equalization apparatus, as shown in fig. 1, includes:
a sound source which is sound information to be reproduced by the system;
the digital signal processor is connected with the output end of the sound source and used for calculating the parameter estimation values of the multi-iteration equalizers at all levels, combining the multi-level equalizers to form a synthesized equalizer and then adding the synthesized equalizer into a signal processing channel;
the power amplifier is connected with the output end of the digital signal processor and used for carrying out power amplification on the equalized signals so as to drive the loudspeaker to produce sound;
a loudspeaker 4 connected with the output end of the power amplifier and used for electro-acoustic conversion so as to reproduce the sound source signal into the air;
and the feedback signal receiving module is connected with the output end of the loudspeaker and is used for collecting and recording the sound signals reproduced into the air.
Further, the sound source is an analog sound source signal generated by various analog devices, or a digital coded signal generated by various digital devices, or a wireless network transmission signal, and the wireless network transmission signal is a broadcast signal transmitted by a wireless transmitting device and is received and demodulated by a wireless receiver to obtain a sound source signal specified by a user. When the sound source is an analog sound source signal, the analog signal is converted into a digital input format specified by a system by an analog-to-digital converter; when the sound source is a digital coding signal, the digital coding signal needs to be converted into a digital input format specified by a system inside a digital signal processor; when the sound source is a wireless network transmission signal, the signal demodulated by the wireless receiver needs to be converted into a digital input format established by a system.
Furthermore, the sound reproduction system has a balanced working mode and a normal playing mode, and the sound source needs to be set and selected according to two different working modes of the sound reproduction system. When the sound reproducing system is placed in an equalization working mode, a sound source is a noise signal generated by a white noise Sequence or a Maximum Length Sequence (MLS) with a specified bandwidth, and the signal presents a flat power spectrum characteristic and is used for training a transfer function from a loudspeaker to a microphone position point in a free field or a reverberation field; when the sound reproduction system is rotated to the normal play operation mode, the sound source is the sound source signal which is specified by the user and needs to be reproduced.
Furthermore, the sound reproduction system has an equalizing operation mode and a normal play mode, and the digital signal processor needs to perform corresponding processing operations according to two different operation modes in which the sound reproduction system is placed. When the sound reproduction system is in an equalization working mode, the digital signal processor firstly carries out channel signal processing on the noise signal to finish the amplitude adjustment and filtering operation of the signal, obtains a transmitting signal suitable for the requirement of the later-stage output bandwidth, then sends the transmitting signal to the power amplifier end, carries out analysis processing on the noise source signal and the feedback signal after receiving the feedback signal, calculates the parameter estimation value of each stage of equalizer according to the iterative LMS criterion, then cascades all the equalizers to obtain the parameter estimation value of the synthesis equalizer, and finally updates the corresponding coefficient value of the FIR filter by using the parameter estimation value of the synthesis equalizer, thereby adding the synthesis equalizer into a signal processing channel; when the sound reproduction system is in a normal playing working mode, the digital signal processor performs equalization processing on an input signal appointed by a user according to the parameter estimation value of the synthesis equalizer and sends the equalized signal to the input end of the power amplifier.
Further, the input interface of the power amplifier can be divided into two types, i.e., a digital input interface and an analog input interface. If the power amplifier is provided with a digital input interface, the power amplifier can directly amplify the digital signal sent by the digital signal processor and then send the digital signal to the loudspeaker end, so that the power amplifier is directly connected with the digital signal processor; if the power amplifier only has an analog input interface, the digital signal sent by the digital signal processor is converted into an analog signal by a digital-to-analog converter, then power amplification is carried out, and finally the analog signal is sent to the loudspeaker end, so that the digital-to-analog converter is connected between the power amplifier and the digital signal processor.
Further, the speaker is not limited to a single speaker unit, the speaker may be implemented as a single speaker unit, or as a speaker array formed by a plurality of speaker units, and the shape of the array may be arranged according to the number of speaker units and the actual application requirement, so as to form various array shapes suitable for the actual application requirement.
Further, the implementation and operation of the feedback signal receiving module need to be determined according to the equalization target of the sound reproduction system. If the sound reproducing system only needs to balance the frequency response fluctuation characteristic of a circuit system consisting of the signal processing part and the power amplifier driving part, the feedback signal receiving module receives and collects the signal at the output end of the power amplifier and sends the collected digital sequence into the digital signal processor; if the sound reproducing system needs the response fluctuation characteristic after the balance circuit part and the loudspeaker part are coupled, the feedback signal receiving module collects the microphone receiving signal which is arranged at a certain expected position point in the space and sends the collected digital sequence to the digital signal processor; if the sound reproducing system needs the response fluctuation characteristic after the balance circuit part, the loudspeaker part and the external environment part are coupled together, the feedback signal receiving module collects the microphone receiving signals which are sequentially arranged at a plurality of spatial position points, and sends the collected digital sequence corresponding to the plurality of spatial position points to the digital signal processor.
Compared with the prior art, the invention has the advantages that:
1. compared with the traditional single LMS or single LPC parameter estimation algorithm, the frequency response equalization method of the sound reproduction system based on online iteration provided by the invention can obviously improve the channel response compensation capability of the synthesis equalizer by increasing the number of the cascade equalizers and increasing the order of each stage of equalizer, so that the overall frequency response curve of the equalized system is more straight, and the synthesis equalizer is close to the response of an ideal inverse filter;
2. the frequency response equalization method of the sound reproduction system of the online iteration can complete parameter estimation and updating processing of the equalizer on line for the environment change of the sound reproduction system and the performance change of the loudspeaker unit. When a user changes the characteristics of a loudspeaker unit and an installation box body of the sound reproduction system or changes the space environment where the sound reproduction system is placed, the system can automatically send out noise signals only by rotating the sound reproduction system to an equalization mode, receives and records feedback signals through a microphone, and completes the tasks of estimating and updating parameters of a synthesis equalizer in real time based on the analysis of the noise and the feedback signals. The response compensation mode based on the online automatic equalization can better meet the actual application requirements, simplifies the equalization operation flow, saves the equalization operation time and has wider and more flexible application;
3. the multi-iteration equalization method provided by the invention can realize more straight equalization processing on the whole broadband internal frequency response expected by a user by increasing the number of iteration equalization, namely increasing the number of cascaded equalizers, and the equalization capability of the multi-iteration equalization method on the broadband internal frequency response is obviously superior to that of the traditional equalization method;
4. the traditional equalizer parameter estimation method needs to obtain the minimum phase response component through the transformation of time domain and frequency domain, and realizes the complex and complicated operation. Compared with the traditional equalizer parameter estimation method, the multiple iteration equalization method provided by the invention directly analyzes the noise signal and the feedback signal in the time domain, and directly completes the parameter estimation of the equalizer in the time domain, and the signal processing flow and hardware are simpler to realize;
5. the invention adopts the iterative algorithm of LMS or NLMS to estimate the parameter value of the equalizer, the parameter estimation method can be completely realized in digital signal processing devices such as DSP and FPGA, the hardware realization is simple, and the cost is lower;
6. the invention generates a single synthesized equalizer by cascading a plurality of estimated equalizers, and in practical application, the single synthesized equalizer is used for executing the equalization operation of the channel, thereby realizing simplicity and reliability.
Drawings
FIG. 1 is a signal processing flow diagram of an on-line iterative audio playback system frequency response equalization method and apparatus of the present invention;
FIG. 2 is a schematic diagram of equalization processing in three different links of a frequency response equalization method and apparatus for an online iterative audio playback system according to the present invention;
fig. 3 is a schematic diagram showing parameter estimation of a multistage equalizer of the frequency response equalization method and apparatus for an on-line iterative sound reproduction system of the present invention, wherein s is a sound source, r is a feedback signal,
the sound source signal is equalized;
FIG. 4 is a schematic diagram of the parameter estimation of the optimal equalizer of level 1 during the 1 st iteration of the present invention, wherein
In order to feed back the signal,in order to be able to detect the error signal,
is a white noise source, and is,
the sound source signal after inverse filtering compensation;
FIG. 5 is a parameter iteration diagram of the level 1 optimal equalizer during the 1 st iteration of the present invention, wherein
In order to feed back the signal,
in order to be able to detect the error signal,
in order to be the step size,
is the equalizer parameter;
FIG. 6 shows the present invention in the first place
In the course of sub-iteration
Parameter estimation scheme for a suboptimal equalizer, wherein
In order to feed back the signal,in order to be able to detect the error signal,
is a white noise source, and is,
the sound source signal after inverse filtering compensation;
FIG. 7 shows the present invention in the first place
In the course of sub-iteration
Parameter iteration schematic diagram of a cascade optimal equalizer, wherein
In order to feed back the signal,in order to be able to detect the error signal,
in order to be the step size,
is the equalizer parameter;
FIG. 8 is a diagram illustrating an implementation of the synthetic equalizer of the present invention, wherein
In order to input the sound source signal,the input signal is processed by the synthesis equalizer;
FIG. 9 is a schematic diagram of the components of an on-line iterative sound reproduction system frequency response equalizer of the present invention;
FIG. 10 is a diagram illustrating a time domain waveform of a noise source signal when the system is operating in an equalization mode according to an embodiment of the present invention;
FIG. 11 is a waveform diagram illustrating a feedback signal received by a microphone when the system is operating in an equalization mode according to an embodiment of the present invention;
fig. 12 is a graph showing a comparison of frequency response curves of the system after equalization is not applied, 1-time iterative equalization is performed, and 10-time iterative equalization is performed in the embodiment of the invention.
Wherein the reference numbers in the figures are:
1. a sound source; 2. a digital signal processor; 3. a power amplifier; 4. a speaker; 5. and a feedback signal receiving module.
Detailed Description
The following detailed description of the preferred embodiments of the present invention, taken in conjunction with the accompanying drawings, will make the advantages and features of the invention easier to understand by those skilled in the art, and thus will clearly and clearly define the scope of the invention.
At present, the traditional equalization method of the sound reproduction system is based on analyzing impulse response functions of the system, fitting a zero-pole model of the impulse response functions, and then finding out the inverse filter response of the system through inversion of the zero-pole, thereby obtaining equalizer parameters of the sound reproduction system. The parameter estimation process of the methods is to calculate the parameters of the equalizer by relying on the single estimation of a least mean square error (LMS) algorithm or a Linear Predictive Coding (LPC) algorithm, the parameters of the inverse filter obtained based on the single estimation method still have certain deviation with the parameters of an ideal inverse filter, and the deviation causes that the frequency response curve of the equalized sound reproduction system still has obvious peak-valley fluctuation in some frequency bands and still does not reach the ideal frequency response flatness characteristic. In order to overcome the defect of certain error in the aspect of system equalizer parameter estimation based on a single LMS or single LPC parameter estimation algorithm, the invention provides a frequency response equalization method and a device of an online iterative sound reproduction system. The invention can obviously improve the response equalization capability of the equalizer by increasing the number of the cascade equalizers, simplifies the equalization operation flow by online automatic equalization, and has wider and more flexible application scenes.
As shown in fig. 9, an on-line iterative sound reproduction system frequency response equalization apparatus according to the present invention is manufactured, and its main body is composed of a sound source 1, a digital signal processor 2, a power amplifier 3, a speaker 4, a feedback signal receiving module 5, and the like.
The sound source 1, when the system is in the rotary equalization mode, the sound source 1 is a white noise signal, the sampling rate is 23.8KHz, the number of bits is 16, and the waveform of the time domain signal is shown in fig. 10. When the system is rotated to the normal play mode, the sound source 1 is a signal to be reproduced designated by the user.
And the digital signal processor 2 is connected with the output end of the sound source 1, and can be realized by taking a DSP or an FPGA as a core processor in hardware implementation. Under the equalization mode, the digital signal processor 2 calculates parameters of each stage of equalizer by combining the noise signal and the feedback signal through a plurality of iterative estimation algorithms, cascades all the equalizers to form a composite equalizer on the basis of finishing the estimation of the multistage equalizer, and realizes the composite equalizer by utilizing an FIR filter. In the normal play mode, the digital signal processor 2 performs equalization processing on the signal to be played back by using a composite equalizer based on the FIR structure.
And the power amplifier 3 is connected with the output end of the digital signal processor 2 and is used for carrying out digital-to-analog conversion and power amplification processing on the digital signal sent by the digital signal processor 2.
And the loudspeaker 4 is connected with the output end of the power amplifier 3, realizes electroacoustic conversion and reproduces sound signals in the air. The loudspeaker 4 is a loudspeaker which has the caliber of 3.5 inches, the rated power of 10 watts and the direct current resistance of 4 ohms and is arranged in the closed box body.
And the feedback signal receiving module 5 is connected with the output end of the loudspeaker 4, and in an equalization mode, the feedback signal receiving module collects a response sequence generated by the excitation of the noise source and sends the response sequence to the digital signal processor 2.
Example (b):
in this embodiment, it is assumed that the sound reproducing system operates in the equalization mode, the speaker is a white noise signal as shown in fig. 10, the microphone is placed 1 meter on the axis of the speaker unit, and the time domain waveform of the feedback signal recorded by the microphone is as shown in fig. 11. Assuming that the order of each stage of equalizer to be estimated is 600, the number of iterative equalization is set to 10.
Fig. 12 shows a comparison of the frequency response curves of the system under three conditions of no equalization, equalization by 1 iteration and equalization by 10 iterations. Comparing the three curves, the system frequency response curve has obvious peak value in the frequency band range of 1.5 KHz-4.5 KHz under the condition of not applying the equalizer; after the 1 st iteration equalization processing, the peak value of the system in the frequency band range of 1.5 KHz-4.5 KHz is eliminated, but the system frequency response curve still has a small amount of fluctuation in the region near the frequency point of 1.5KHz, and meanwhile, the system frequency response curve still has a large degree of fluctuation in the frequency band of 100 Hz-200 Hz; after 10 times of iterative equalization processing, a small amount of fluctuation of the system in the area near the 1.5KHz frequency point is eliminated, and meanwhile, the amplitude peak value in the frequency band of 100 Hz-200 Hz is also inhibited to a greater degree. Comparing the frequency response curves after the 1 st iterative equalization and the 10 th iterative equalization, it can be seen that: by increasing the iteration times, the iterative equalization method provided by the invention can obviously improve the flatness degree of the frequency response curve after the system equalization, which shows that the multiple iterative equalization method provided by the invention has better equalization effect compared with the traditional equalization method, and the frequency response curve after the equalization is more flat.
The above embodiments are merely illustrative of the technical ideas and features of the present invention, and the purpose thereof is to enable those skilled in the art to understand the contents of the present invention and implement the present invention, and not to limit the protection scope of the present invention. All equivalent changes and modifications made according to the spirit of the present invention should be covered within the protection scope of the present invention.
Claims (14)
1. A frequency response equalization method for an online iterative sound reproduction system comprises the following steps:
(1) setting a sound source as a noise signal in a system rotary equalization mode, then controlling the system to play the noise signal, and simultaneously collecting and recording a feedback signal by a sensor;
(2) combining the noise signal and the feedback signal, and sequentially estimating parameters of each stage of equalizer by using a step-by-step iterative minimum mean square error criterion;
(3) cascading the equalizers at all levels to form a synthesized equalizer;
(4) and placing the synthesized equalizer in a signal processing channel to finish the frequency response equalization operation of the system, and then placing the system in a normal play mode.
2. The on-line iterative audio playback system frequency response equalization method of claim 1, characterized in that: the noise signal in the step (1) is a noise signal generated by a white noise sequence or a maximum length sequence with a specified bandwidth.
3. The on-line iterative audio playback system frequency response equalization method of claim 1, characterized in that: in the step (2), the parameters of the equalizers at each stage are sequentially estimated by using the minimum mean square error criterion of the step-by-step iteration, and the parameter estimation process of the equalizers at each stage based on the minimum mean square error criterion of the step-by-step iteration is as follows:
a. assume that the time domain sequence vector of the input noise source signal is:
wherein,
the number of sampling points of a time domain discrete sequence of a noise source signal; assuming that the time domain impulse response sequence expression of the acoustic playback system is:
，
wherein,
is the sequence length of the system time domain impulse response; suppose that the time domain sequence of the feedback signal collected by the sensor is:
，
wherein ""represents a convolution operation between two sequence vectors; in the 1 st iteration, the length of the time domain impulse response sequence of the 1 st equalizer to be solved is assumed to be
The response sequence vector is then expressed as:
feedback signal
Via the 1 st equalizer to be solved
After processing, the expression of the feedback signal after 1 st equalization is obtained as follows:
，
combined with sound source signals
And the 1 st equalized feedback signal
Calculating the sound source signal according to the minimum mean square error criterion
And the equalized feedback signal
The equalizer parameter vector when the mean square error between the two is the minimum is used as the estimated value of the 1 st optimal equalizer parameter vector designed in the 1 st iteration, and the expression is as follows:
calculating an estimated value of an optimal equalizer parameter vector according to a minimum mean square error algorithm, wherein the calculation of the optimal equalizer parameter is performed by an iterative algorithm based on the minimum mean square error algorithm
In the secondary iteration process, the calculation expression of the 1 st optimal equalizer parameter is as follows:
wherein
Representing sound source signals
The delay amount of (3); from the above expression, the parameter iteration process of the 1 st optimal equalizer is shown, wherein
The step length is the length of the step, and the value of the step length satisfies the following relation:
wherein
Is a matrix
The trace of (a) is determined,
，
is a vector
The conjugate of (2) transposes the vector, and
is a length of feedback signal received by the sensor
In the second place of
Feedback signal of sampling timeThe vector expression of (a) is:
；
step sizeAccording to the followingSelecting from the surface relations:，
wherein
Is a matrixThe maximum eigenvalue of (d);
the assumption is made by the minimum mean square error algorithm
The algorithm tends to converge after the iterative computation, and the parameter estimation value of the 1 st equalizer is as follows:
estimation of parameters for the 1 st equalizer
Carrying out normalization processing to obtain:
wherein,
is a vector
The transposed vector of (1);
feedback signal
Via the 1 st equalizerAfter processing, the expression of the feedback signal after 1 st equalization is obtained as follows:
combining the 1 st equalized feedback signals
And sound source signal
Calculating the root mean square error value between the feedback signal processed by the 1 st stage equalizer and the sound source signal
The expression is as follows:
，
b. according to the estimation process of the equalizer parameters in the step a, continuously completing the 1 st, 2 nd, … th,
Estimation of parameters of the equalizer, assuming that
Based on the equalizer parameter estimation, the
The estimation procedure for each equalizer parameter is as follows:
before menstruation
Feedback signal after filtering processing of equalizer
Expressed as:
before menstruation
Feedback signal after filtering processing of equalizer
Then via the first one to be askedEqualizer
After treatment, a pre-treatment is obtained
The expression of the feedback signal after the equalizer processing is as follows:
，
combined with sound source signals
And the warpFeedback signal processed by equalizerCalculating the sound source signal according to the minimum mean square error criterion
And the equalized feedback signal
The equalizer parameter vector when the mean square error between them is minimum is taken as the second
Design at time of sub-iteration
The estimated value of the optimal equalizer parameter vector is expressed as:
,
calculating an estimated value of an optimal equalizer parameter vector according to a minimum mean square error algorithm, wherein the calculation of the optimal equalizer parameter is performed by an iterative algorithm based on the minimum mean square error algorithm
In the course of the second iteration, the
The computational expression for each optimal equalizer parameter is as follows:
,
according to the above expression, the first
Iterative process of equalizer parameters, in which the step size is
The values of (A) satisfy the following relations:
，
is a vector
The conjugate of (2) transposes the vector, and
is before menstruationThe length of the feedback signal after the filtering processing of the equalizer is
In the second place of
One sampling time is before
The vector expression of the time domain of the feedback signal after the filtering processing of the optimal equalizer is as follows:；
step size
Selected according to the following relationship:
，
wherein
Is a matrix
The maximum eigenvalue of (d);
the assumption is made by the minimum mean square error algorithmThe algorithm tends to converge after the sub-iterative computation, when the first time
The parameter estimates for each equalizer are:
to the second
Estimation of equalizer parameters
Carrying out normalization processing to obtain:
wherein,
is a vector
The transposed vector of (1); before menstruation
Feedback signal after filtering processing of equalizerThen via the first
Equalizer
After treatment, a pre-treatment is obtained
The feedback signal after the equalizer processing has the expression:
，
combined with premenstrual syndrome
Feedback signal processed by equalizer
And sound source signalCalculating the longitudeRoot mean square error value between feedback signal and sound source signal after stage equalizer processing
The expression is as follows:
c. according to the estimation process of the equalizer parameter in step b, continuously completing the second step、
、…、
Parameter estimation of an equalizer, in the first place
After parameter estimation of the equalizer, the root error value is equalized
Less than the desired root mean square error set by the user
And the algorithm does not continue to calculate the parameters of the next-stage equalizer, and the iteration is stopped.
4. The on-line iterative audio playback system frequency response equalization method of claim 3, characterized in that: during iterative calculations to estimate equalizer parameters, the algorithm always monitors the equalization root error value
After each iteration is completed, the root mean square of the current time interval is equal to the expected Root Mean Square (RMS) set by the userError of the measurement
Making a comparison and controlling the operation of the iteration loop, if the next iteration starts
Then the next iteration continues, if
The iteration is stopped.
5. The on-line iterative audio playback system frequency response equalization method of claim 3, characterized in that: in the step a, the optimal equalizer parameter can be calculated by using a normalized minimum mean square error algorithm, namely
In the secondary iteration process, the 1 st optimal equalizer parameter calculation expression based on the normalized minimum mean square error algorithm is as follows:
wherein
For the purpose of the step-size factor,
is a small positive constant number of the cells,
。
6. the online iteration of claim 1The frequency response equalization method of the sound reproduction system is characterized by comprising the following steps: in the step a, the optimal equalizer parameter can be calculated by using a normalized minimum mean square error algorithm, namely
In the process of sub-iteration, based on the normalized minimum mean square error algorithmThe computational expression for each optimal equalizer parameter is as follows:
wherein
For the purpose of the step-size factor,
is a small positive constant number of the cells,
。
7. the on-line iterative audio playback system frequency response equalization method of claim 1, characterized in that: the equalizer of each stage is cascaded in the step (3) to form a synthesized equalizer, and the implementation process is as follows:
by
This obtained by sub-iterative estimation
The parameter estimation values of the equalizers are respectively:
、
、…、
from this
Composite equalizer formed by cascading equalizers
The expression of (a) is:
wherein ""represents a convolution operation between time domain vector sequences.
8. The on-line iterative audio playback system frequency response equalization method of claim 1, characterized in that: the step (4) of placing the synthesized equalizer in the signal processing channel to complete the system frequency response equalization operation is specifically implemented as follows:
in obtaining a composite equalizerBased on the above, the finite impulse response filter is used to implement the equalization operation, and it is assumed that the time domain sequence vector of the input sound source signal is:
synthesized equalizer
The processed sound source signal may be represented as:
the sound source signal processed by the synthesis equalizer is amplified by a power amplifier and then sent to a loudspeaker end to drive the loudspeaker to radiate sound waves.
9. The on-line iterative audio playback system frequency response equalization apparatus of claim 1, wherein: the signal processing channel in the step (4) is to complete the amplitude adjustment and filtering operation of the signal in the digital signal processor to obtain a transmission signal suitable for the requirement of the later-stage output bandwidth.
10. An on-line iterative sound reproduction system frequency response equalization device is characterized in that: the device comprises a sound source (1), a digital signal processor (2) which is connected with the output end of the sound source (1) and used for calculating the parameter estimation values of various levels of equalizers of multiple iterations and combining the multiple levels of equalizers to form a synthesized equalizer and then adding the synthesized equalizer into a signal processing channel, a power amplifier (3) which is connected with the output end of the digital signal processor (2) and used for amplifying the power of the equalized signal to drive a loudspeaker to sound, the device comprises a power amplifier (3), a loudspeaker (4) connected with the output end of the power amplifier (3) and used for electro-acoustic conversion so as to replay a sound source signal into the air, and a feedback signal receiving module (5) connected with the output end of the loudspeaker (4) and used for collecting and recording the sound signal replayed into the air, wherein the sound source is the sound information to be replayed by the system.
11. The on-line iterative audio playback system frequency response equalization apparatus of claim 10, wherein: the sound source (1) is an analog sound source signal generated by various analog devices, or a digital coding signal generated by various digital devices, or a wireless network transmission signal, and the wireless network transmission signal is a broadcast signal transmitted by a wireless transmitting device and is received and demodulated by a wireless receiver to obtain a sound source signal appointed by a user.
12. The on-line iterative audio playback system frequency response equalization apparatus of claim 10, wherein: the sound reproduction system is provided with a balance working mode and a normal play working mode, when the sound reproduction system is in the balance working mode, the sound source (1) is a noise signal generated by a white noise sequence or a maximum length sequence with specified bandwidth, and when the sound reproduction system is in the normal play working mode, the sound source (1) is a sound source signal which is specified by a user and needs to be reproduced.
13. The on-line iterative audio playback system frequency response equalization apparatus of claim 10, wherein: when the power amplifier (3) is provided with a digital input interface, the power amplifier (3) is directly connected with the digital signal processor (2); when the power amplifier (3) only has an analog input interface, a digital-to-analog converter for converting the digital signal sent by the digital signal processor (2) into an analog signal is connected between the power amplifier (3) and the digital signal processor (2).
14. The on-line iterative audio playback system frequency response equalization apparatus of claim 10, wherein: the loudspeaker (4) is a single loudspeaker unit or a loudspeaker array composed of a plurality of loudspeaker units and can be arranged according to the number of the loudspeaker units and the actual application requirement.
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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 |
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