CN112666521A - Indoor sound source positioning method based on improved self-adaptive notch filter - Google Patents
Indoor sound source positioning method based on improved self-adaptive notch filter Download PDFInfo
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Abstract
The invention belongs to the technical field of navigation positioning, and discloses an indoor sound source positioning method based on an improved self-adaptive wave trap, which comprises the steps of obtaining indoor height information and obtaining relative azimuth information of a sound source according to a quaternary microphone array arranged indoors; constructing a sound source position information equation according to indoor height information and relative azimuth information of a sound source, and taking a first two-array element phase difference and a second two-array element phase difference as undetermined values of the sound source position information equation; and adding a fast Fourier transform module and an inverse fast Fourier transform module into the adaptive notch filter to obtain an improved adaptive notch filter, calculating by using the improved adaptive notch filter to obtain the exact values of the phase difference between the first two array elements and the phase difference between the second two array elements, and substituting the exact values into a sound source position information equation to obtain the three-dimensional position information of the sound source. The invention solves the problem of lower indoor sound source positioning precision in the prior art, and can improve the indoor sound source positioning precision.
Description
Technical Field
The invention relates to the technical field of navigation positioning, in particular to an indoor sound source positioning method based on an improved self-adaptive wave trap.
Background
In the times that smart cities with high intelligentization and informatization characteristics are gradually favored by people, indoor positioning becomes an indispensable component of smart cities. Although a Global Navigation Satellite System (GNSS) has a positioning accuracy of cm level, since satellite signals cannot penetrate through a building, the signals are seriously weakened in an indoor environment, and a traditional outdoor positioning mode cannot meet the requirements of people on indoor Navigation, positioning and tracking.
One of the methods used in indoor positioning is sound source positioning of a microphone array, wherein sound source positioning of the microphone array is realized by using an improved cross-power spectrum phase method, and the error precision of distance positioning is less than +/-20 cm. Based on the improved PHAT-GCC method, the noise sharpening peak can be effectively inhibited under the condition of low signal to noise ratio, so that the sound source positioning accuracy is improved. However, the signal sources in the above manner are all broadband signals. Therefore, the above methods all adopt a generalized cross-correlation time delay estimation method, belong to a sound source positioning mode of broadband signals, and have the disadvantage that the method is strongly interfered by reverberation and noise, and if the signal noise is not properly processed, the positioning accuracy is greatly reduced.
Disclosure of Invention
The invention solves the problem of low indoor sound source positioning precision in the prior art by providing an improved adaptive notch filter-based indoor sound source positioning method.
The invention provides an indoor sound source positioning method based on an improved self-adaptive notch filter, which comprises the following steps:
step 3, adding a fast Fourier transform module and an inverse fast Fourier transform module into the adaptive notch filter to obtain an improved adaptive notch filter, calculating by using the improved adaptive notch filter to obtain the exact values of the phase difference between the first two array elements and the phase difference between the second two array elements, and substituting the exact values into the sound source position information equation to obtain the three-dimensional position information of the sound source;
the improved self-adaptive notch filter adopts fast Fourier transform to transform a sound source signal acquired by the quaternary microphone array into a frequency domain signal, the frequency domain signal is filtered to obtain a frequency domain filtering signal, the frequency domain filtering signal is subjected to inverse fast Fourier transform to obtain a time domain signal, and the time domain signal is used as an exact value of the first two-array element phase difference and the second two-array element phase difference.
Preferably, in the step 1, the indoor height information is obtained through laser ranging; the carrier frequency f corresponding to the sound source meets the following conditions: f is less than c/2 d; where c denotes the sound velocity and d denotes the spacing between two microphones located on the diagonal in the quaternary microphone array.
Preferably, in step 1, the specific implementation manner of obtaining the relative azimuth information of the sound source according to the quaternary microphone array arranged indoors is as follows:
acquiring a waveform diagram of four paths of receiving signals corresponding to the quaternary microphone array; obtaining phase difference information among the four paths of received signals according to the oscillogram; and obtaining the relative azimuth information of the sound source according to the phase difference information.
Preferably, in step 2, the sound source position information equation is as follows:
x＝Rcosα
y＝Rcosβ
in the formula, R is the distance from a sound source position S to a central point O of a quaternary microphone array, a coordinate system (x, y, z) of an O-ABCD carrier of the quaternary microphone array is set as a system coordinate system, the direction from an array element A to an array element C on the horizontal plane of the quaternary microphone array is taken as the negative direction of an x axis, the direction from an array element B to an array element D is taken as the negative direction of a y axis, the central point of the quaternary microphone array is taken as the positive direction of a z axis vertically downwards, and the x, the y and the z are respectively the coordinate values of the x axis, the coordinate value of; alpha is a radial vector between the quaternary microphone array center point O and the sound source position S
The angle with the x-axis; beta is a radial vector
The included angle with the y axis; t is the sound wave arrival time; and c is the propagation speed of the sound wave in the medium.
Preferably, in the step 2, a first coefficient is set
Second coefficient of
Then there are:
z＝h (3)
in the formula, d is the distance between two array elements on the coaxial line; λ is the wavelength of the sound source; h is the vertical height of the sound source from the central point of the quaternary microphone array;
the phase difference between the A array element and the C array element is recorded as the first two array elementsPhase difference;
the phase difference between the B array element and the D array element is recorded as the phase difference between the second two array elements.
Preferably, in step 3, the specific implementation manner of calculating the exact values of the first two-array element phase difference and the second two-array element phase difference by using the improved adaptive notch filter is as follows:
estimating the deviation degree of the signal frequency and the quantization frequency point, and carrying out frequency shift on the sound source signal acquired by the quaternary microphone array according to the deviation degree; the signal frequency is the maximum frequency in the frequency domain signals obtained after fast Fourier transform is carried out on the sound source signals, the quantization frequency point is the actual signal frequency obtained, and band-pass filtering is carried out on non-signal frequencies around the frequency domain signals obtained by the sound source signals to obtain noise-reduction sound source signals without other interference frequencies;
carrying out frequency domain filtering on the frequency-shifted signal, and carrying out inverse fast Fourier transform on the processed frequency domain filtering signal to reconstruct a time domain signal;
and performing correlation calculation according to the time domain signals after noise reduction to obtain the phase deviation of the received signals of each microphone, wherein the phase deviations are obtained by performing cross-correlation operation on the same array element, performing self-adaptive updating on the first coefficient and the second coefficient by using a self-adaptive algorithm, and performing position coordinate calculation to obtain the exact values of the phase difference between the first two array elements and the phase difference between the second two array elements.
Preferably, the input signal of the improved adaptive notch filter is represented as:
where x (k) is the received signal value formed by different discrete point values of the received signal, AxTo receive the amplitude of the signal, wk is the frequency of the input signal,
the signal phase deviation value of the receiving array element relative to the reference array element is obtained; n iskK is a discrete value of an acquired signal, and if m points are acquired in total to form an input signal, k represents from 1 to m;
from the cosine function properties:
after the input signal is noise-filtered, removing the noise value nkObtaining:
the output signal of the improved adaptive notch filter is represented as:
y(k)＝ws(k)rs(k)+wc(k)rc(k) (5)
in the formula, rs(k)、rc(k) Respectively sine reference signal, cosine reference signal, ws(k)、wc(k) The weight coefficients of the sine reference signal and the cosine reference signal are respectively, and omega represents the central frequency of the reference signal; r iss(k)＝Axsinwk、rc(k)＝Axcoswk、
The error formula of the improved adaptive filter is as follows:
e(k)＝x(k)-y(k) (6)
updating the weight coefficient of the sine reference signal and the weight coefficient of the cosine reference signal according to the feedback error value e (k):
ws(k+1)＝ws(k)+μe(k)rs(k) (7)
wc(k+1)＝wc(k)+μe(k)rc(k) (8)
where μ is the adaptive step size, e (k) is the error offset between the input signal and the iterated output result, ws(k +1) is the weight coefficient of the updated sinusoidal reference signal, wc(k +1) is the weight coefficient of the cosine reference signal after updating,
the phase estimation value in the actual operation process is obtained;
estimate the phase
Substituted into formula (4)
And obtaining a new fitting signal, then continuously iterating to obtain a phase estimation value until obtaining the phase offset caused by the actual sound source propagation delay, and when the error between the array received signal and the replica signal is smaller than a preset range, taking the correspondingly obtained phase estimation value as the phase offset caused by the actual delay to finally obtain the delay time.
Preferably, the adaptive step size is determined by the following formula:
wherein f issFor receiving the sampling frequency of the acoustic signal emitted by the source, BfFor full bandwidth, AxIs the amplitude of the input signal.
Preferably, the phase difference between the two array elements is obtained by subtracting the output signals y (k) of the two improved adaptive notch filters, and the following formula is adopted for calculation:
in the formula (I), the compound is shown in the specification,
representing the phase difference between two array elements including the first two-array element phase difference
Second two-array element phase difference
Representing the phase of a microphone element,
Representing the phase of microphone element two; w is as1Weight coefficients, w, representing sinusoidal reference signals derived from a microphone element and a received signal ss2Weight coefficient, w, representing a sinusoidal reference signal derived on the basis of a microphone element number two and a received signal sc1Weight coefficient, w, representing a cosine reference signal derived on the basis of a microphone element and a received signal sc2Representing the weighting coefficients of a cosine reference signal derived on the basis of the microphone element number two and the received signal s.
One or more technical schemes provided by the invention at least have the following technical effects or advantages:
in the invention, the provided indoor sound source positioning method based on the improved adaptive notch filter improves the traditional adaptive notch filter, integrates a frequency domain noise reduction method based on FFT (fast Fourier transform) narrowband signals, carries out frequency shift on the signals according to the deviation degree of the estimated signal frequency from a quantized frequency point, carries out frequency domain filtering, and carries out IFFT (inverse fast Fourier transform) on the processed frequency domain filtering signals to reconstruct time domain signals. The narrow-band signal frequency domains are concentrated in one region, and the noise is uniformly distributed in each part of the frequency domains, so that the noise outside the signal frequency domain is effectively inhibited, the indoor sound source positioning accuracy is improved, and the method has important significance for the actual indoor sound source positioning method.
Drawings
Fig. 1 is a flowchart of an indoor sound source localization method based on an improved adaptive notch filter according to an embodiment of the present invention;
fig. 2 is a diagram of a quaternary positioning model in an indoor sound source positioning method based on an improved adaptive notch filter according to an embodiment of the present invention;
fig. 3 is a schematic structural diagram of an improved adaptive notch filter used in an indoor sound source localization method based on an improved adaptive notch filter according to an embodiment of the present invention;
fig. 4 is a first graph of delay relationships between microphone received signals in an indoor sound source localization method based on an improved adaptive notch filter according to an embodiment of the present invention;
fig. 5 is a diagram showing a delay relationship between microphone received signals in an indoor sound source localization method based on an improved adaptive notch filter according to an embodiment of the present invention;
fig. 6 is a graph comparing error means values of an improved adaptive notch filter and a conventional notch filter, which are used in an indoor sound source localization method based on the improved adaptive notch filter according to an embodiment of the present invention.
Detailed Description
In order to better understand the technical solution, the technical solution will be described in detail with reference to the drawings and the specific embodiments.
The invention provides an indoor sound source positioning method based on an improved adaptive notch filter, which comprises the following steps as shown in figure 1:
Specifically, the indoor height information is obtained through laser ranging; the carrier frequency f corresponding to the sound source meets the following conditions: f is less than c/2 d; where c denotes the sound velocity and d denotes the spacing between two microphones located on the diagonal in the quaternary microphone array.
The concrete implementation mode of obtaining the relative azimuth information of the sound source according to the quaternary microphone array arranged indoors is as follows: acquiring a waveform diagram of four paths of receiving signals corresponding to the quaternary microphone array; obtaining phase difference information among the four paths of received signals according to the oscillogram; and obtaining the relative azimuth information of the sound source according to the phase difference information.
And 2, constructing a sound source position information equation according to the indoor height information and the relative azimuth information of the sound source, and taking the phase difference of the first two array elements and the phase difference of the second two array elements as undetermined values of the sound source position information equation.
Specifically, the sound source position information equation is shown as follows:
x＝Rcosα
y＝Rcosp
in the formula, R is the distance from a sound source position S to a central point O of a quaternary microphone array, a coordinate system (x, y, z) of an O-ABCD carrier of the quaternary microphone array is set as a system coordinate system, the direction from an array element A to an array element C on the horizontal plane of the quaternary microphone array is taken as the negative direction of an x axis, the direction from an array element B to an array element D is taken as the negative direction of a y axis, the central point of the quaternary microphone array is taken as the positive direction of a z axis vertically downwards, and the x, the y and the z are respectively the coordinate values of the x axis, the coordinate value of; alpha is a radial vector between the quaternary microphone array center point O and the sound source position S
The angle with the x-axis; beta is a radial vector
The included angle with the y axis; t is the sound wave arrival time; and c is the propagation speed of the sound wave in the medium.
Set the first coefficient
Second coefficient of
Then there are:
z＝h (3)
in the formula, d is the distance between two array elements on the coaxial line; λ is the wavelength of the sound source; h is the vertical height of the sound source from the central point of the quaternary microphone array;
the phase difference between the array element A and the array element C is recorded as the phase difference between the first two array elements;
the phase difference between the B array element and the D array element is recorded as the phase difference between the second two array elements.
And 3, adding a fast Fourier transform module and an inverse fast Fourier transform module into the adaptive notch filter to obtain an improved adaptive notch filter, calculating by using the improved adaptive notch filter to obtain the exact values of the phase difference between the first two array elements and the phase difference between the second two array elements, and substituting the exact values into the sound source position information equation to obtain the three-dimensional position information of the sound source.
The improved self-adaptive notch filter adopts fast Fourier transform to transform a sound source signal acquired by the quaternary microphone array into a frequency domain signal, the frequency domain signal is filtered to obtain a frequency domain filtering signal, the frequency domain filtering signal is subjected to inverse fast Fourier transform to obtain a time domain signal, and the time domain signal is used as an exact value of the first two-array element phase difference and the second two-array element phase difference.
Specifically, the specific implementation manner of obtaining the exact values of the first two-array element phase difference and the second two-array element phase difference by using the improved adaptive notch filter is as follows: estimating the deviation degree of the signal frequency and the quantization frequency point, and carrying out frequency shift on the sound source signal acquired by the quaternary microphone array according to the deviation degree; the signal frequency is the maximum frequency in the frequency domain signals obtained after fast Fourier transform is carried out on the sound source signals, the quantization frequency point is the actual signal frequency obtained, and band-pass filtering is carried out on non-signal frequencies around the frequency domain signals obtained by the sound source signals to obtain noise-reduction sound source signals without other interference frequencies; carrying out frequency domain filtering on the frequency-shifted signal, and carrying out inverse fast Fourier transform on the processed frequency domain filtering signal to reconstruct a time domain signal; and performing correlation calculation according to the time domain signals after noise reduction to obtain the phase deviation of the received signals of each microphone, wherein the phase deviations are obtained by performing cross-correlation operation on the same array element, performing self-adaptive updating on the first coefficient and the second coefficient by using a self-adaptive algorithm, and performing position coordinate calculation to obtain the exact values of the phase difference between the first two array elements and the phase difference between the second two array elements.
The input signal of the improved adaptive notch filter is represented as:
where x (k) is the received signal value formed by different discrete point values of the received signal, AxTo receive the amplitude of the signal, wk is the frequency of the input signal,
the signal phase deviation value of the receiving array element relative to the reference array element is obtained; n iskK is a discrete value of an acquired signal, and if m points are acquired in total to form an input signal, k represents from 1 to m;
from the cosine function properties:
after the input signal is noise-filtered, removing the noise value nkObtaining:
the output signal of the improved adaptive notch filter is represented as:
y(k)＝ws(k)rs(k)+wc(k)rc(k) (5)
in the formula, rs(k)、rc(k) Respectively sine reference signal, cosine reference signal, ws(k)、wc(k) The weight coefficients of the sine reference signal and the cosine reference signal are respectively, and omega represents the central frequency of the reference signal; r iss(k)＝Axsinwk、rc(k)＝Axcoswk、
The error formula of the improved adaptive filter is as follows:
e(k)＝x(k)-y(k) (6)
updating the weight coefficient of the sine reference signal and the weight coefficient of the cosine reference signal according to the feedback error value e (k):
ws(k+1)＝ws(k)+μe(k)rs(k) (7)
wc(k+1)＝wc(k)+μe(k)rc(k) (8)
where μ is the adaptive step size, e (k) is the error offset between the input signal and the iterated output result, ws(k +1) is the updated sinusoidal reference signalWeight coefficient of (d), wc(k +1) is the weight coefficient of the cosine reference signal after updating,
is the phase estimation value in the actual operation process.
Estimate the phase
Substituted into formula (4)
And obtaining a new fitting signal, then continuously iterating to obtain a phase estimation value until obtaining the phase offset caused by the actual sound source propagation delay, and when the error between the array received signal and the replica signal is smaller than a preset range, taking the correspondingly obtained phase estimation value as the phase offset caused by the actual delay to finally obtain the delay time.
The adaptation step size is determined by:
wherein f issFor receiving the sampling frequency of the acoustic signal emitted by the source, BfFor full bandwidth, AxIs the amplitude of the input signal.
Obtaining the phase difference of two array elements by taking the difference of output signals y (k) of the two improved adaptive notch filters, wherein the calculation adopts the following formula:
in the formula (I), the compound is shown in the specification,
representing the phase difference between two array elements including the first two-array element phase difference
Second two-array element phase difference
Representing the phase of a microphone element,
Representing the phase of microphone element two; w is as1Weight coefficients, w, representing sinusoidal reference signals derived from a microphone element and a received signal ss2Weight coefficient, w, representing a sinusoidal reference signal derived on the basis of a microphone element number two and a received signal sc1Weight coefficient, w, representing a cosine reference signal derived on the basis of a microphone element and a received signal sc2Representing the weighting coefficients of a cosine reference signal derived on the basis of the microphone element number two and the received signal s.
The present invention will be described in addition below.
The embodiment provides an indoor sound source positioning method based on an improved adaptive notch filter, which comprises the following steps:
and step S1, obtaining indoor height information Z in advance by laser ranging in a room where the quaternary microphone array is arranged, and determining the relative azimuth of the sound source according to the phase difference between the signals received by the measuring and comparing microphones after the microphones receive four paths of signals (A, B, C, D four azimuths).
The step S1 includes the following sub-steps:
and step S11, using a quaternary microphone array of ReSpeaker 4-Mics Pi HAT as a sound source receiving array, and using a raspberry RaspberryPi-3b + as a core processing system to realize the indoor positioning device. The practical application range of the quaternary microphone array distance d is between 5cm and 10 cm.
In step S12, at room temperature, if the sound velocity c is about 340m/S, the signal carrier frequency f is set to f < c/2d, and the indoor height Z is measured as a determination value by laser ranging.
Step S13, according to the signal characteristics, a waveform diagram of the four-path signals under different signal-to-noise ratios (SNRs) is drawn, and according to the phase shift relationship of the waveforms, the delay relationship between the received signals of the four-path microphones can be known, so as to obtain the relative orientation, as shown in fig. 4 and 5.
Step S2, establishing a sound source position information equation by using height and phase information according to the data obtained in the step 1, and further determining a pending coefficient A, C phase difference
Out of phase with B, D
The step S2 includes the following sub-steps:
step S21, setting a coordinate system of a microphone array (O-ABCD) carrier as a system coordinate system (x, y, z), as shown in fig. 2, establishing a sound source position information equation according to the height and phase information obtained in step 1, and setting coefficients
Coefficient of performance
Then there are:
z＝h (3)
in the formula, d is the distance between two array elements on the coaxial line; λ is the wavelength of the sound source;
is the phase difference between the A, C array elements,
is the phase difference between B, D array elements.
Step S22, determining the undetermined value in the A, B coefficient through the sound source position information equation established in step S21
And
step S3, pending value
And
and (3) calculating by using an improved self-adaptive notch filter to obtain an exact value, and replacing the exact value into the sound source positioning information equation in the step (2), so that the three-dimensional position information of the sound source position in the indoor environment can be obtained.
The step S3 includes the following sub-steps:
in step S31, the frequency of the signal is shifted according to the estimated deviation degree of the signal frequency from the quantization frequency point.
Step S32, frequency-domain filtering is performed on the frequency-shifted signal, and IFFT is performed on the processed frequency-domain filtered signal to reconstruct a time-domain signal.
Step S33, after the above processing, adaptively updates the trap coefficients by using an adaptive Least Mean Square (LMS) algorithm, and the whole structure is as shown in fig. 3.
Step S33, calculating the undetermined coefficient by the improved adaptive notch filter method
And
and substituting the indoor positioning three-dimensional coordinate information into the sound source positioning information equation in the step 2 to obtain the indoor positioning three-dimensional coordinate information.
Fig. 6 is a graph comparing error averages of an improved adaptive notch filter and a conventional notch filter, which are used in an indoor sound source localization method based on the improved adaptive notch filter according to an embodiment of the present invention. As can be seen from fig. 6, the mean value of the error of the improved adaptive notch filter adopted by the present invention is smaller than that of the conventional notch filter, which indicates that the present invention can effectively improve the accuracy of positioning the indoor sound source.
The indoor sound source positioning method based on the improved adaptive notch filter provided by the embodiment of the invention at least comprises the following technical effects:
(1) in the traditional array signal positioning, the improved adaptive notch filter is adopted, and the frequency domain noise reduction method based on the FFT narrowband signal is integrated by the improved adaptive notch filter, so that the situation that the positioning precision is reduced due to the fact that broadband signals are easily interfered by reverberation and noise is avoided, the influence of the noise can be effectively reduced, and the positioning precision of an indoor sound source is improved.
(2) The invention solves the problem of large computation amount and incapability of realizing real-time performance in the MUSIC method or the ESPRIT method adopted in the traditional array signal positioning, solves the narrow-band signal through the improved adaptive notch filter, obtains a phase estimation value only through simple signal iteration and addition and subtraction judgment, and does not add complex matrix operation with the traditional MUSIC method or the ESPRIT method, thereby realizing indoor positioning with small computation amount and high real-time performance.
Finally, it should be noted that the above embodiments are only for illustrating the technical solutions of the present invention and not for limiting, and although the present invention has been described in detail with reference to examples, it should be understood by those skilled in the art that modifications or equivalent substitutions may be made on the technical solutions of the present invention without departing from the spirit and scope of the technical solutions of the present invention, which should be covered by the claims of the present invention.
Claims (9)
1. An indoor sound source positioning method based on an improved adaptive notch filter is characterized by comprising the following steps:
step 1, acquiring indoor height information, and acquiring relative azimuth information of a sound source according to a quaternary microphone array arranged indoors;
step 2, constructing a sound source position information equation according to the indoor height information and the relative azimuth information of the sound source, and taking the phase difference of the first two array elements and the phase difference of the second two array elements as undetermined values of the sound source position information equation;
step 3, adding a fast Fourier transform module and an inverse fast Fourier transform module into the adaptive notch filter to obtain an improved adaptive notch filter, calculating by using the improved adaptive notch filter to obtain the exact values of the phase difference between the first two array elements and the phase difference between the second two array elements, and substituting the exact values into the sound source position information equation to obtain the three-dimensional position information of the sound source;
the improved self-adaptive notch filter adopts fast Fourier transform to transform a sound source signal acquired by the quaternary microphone array into a frequency domain signal, the frequency domain signal is filtered to obtain a frequency domain filtering signal, the frequency domain filtering signal is subjected to inverse fast Fourier transform to obtain a time domain signal, and the time domain signal is used as an exact value of the first two-array element phase difference and the second two-array element phase difference.
2. The improved adaptive notch filter based indoor sound source positioning method according to claim 1, wherein in the step 1, height information of the indoor space is obtained by laser ranging; the carrier frequency f corresponding to the sound source meets the following conditions: f is less than c/2 d; where c denotes the sound velocity and d denotes the spacing between two microphones located on the diagonal in the quaternary microphone array.
3. The method for positioning an indoor sound source based on an improved adaptive notch filter according to claim 1, wherein in the step 1, the specific implementation manner of obtaining the relative azimuth information of the sound source according to the quaternary microphone array arranged indoors is as follows:
acquiring a waveform diagram of four paths of receiving signals corresponding to the quaternary microphone array; obtaining phase difference information among the four paths of received signals according to the oscillogram; and obtaining the relative azimuth information of the sound source according to the phase difference information.
4. The improved adaptive notch filter based indoor sound source positioning method according to claim 1, wherein in the step 2, the sound source position information equation is as follows:
x＝Rcosα
y＝Rcosβ
in the formula, R is the distance from a sound source position S to a central point O of a quaternary microphone array, a coordinate system (x, y, z) of an O-ABCD carrier of the quaternary microphone array is set as a system coordinate system, the direction from an array element A to an array element C on the horizontal plane of the quaternary microphone array is taken as the negative direction of an x axis, the direction from an array element B to an array element D is taken as the negative direction of a y axis, the central point of the quaternary microphone array is taken as the positive direction of a z axis vertically downwards, and the x, the y and the z are respectively the coordinate values of the x axis, the coordinate value of; alpha is a radial vector between the quaternary microphone array center point O and the sound source position S
The angle with the x-axis; beta is a radial vector
The included angle with the y axis; t is the sound wave arrival time; and c is the propagation speed of the sound wave in the medium.
5. The method as claimed in claim 4, wherein in step 2, a first coefficient is set
Second coefficient of
Then there are:
z＝h (3)
in the formula, d is the distance between two array elements on the coaxial line; λ is the wavelength of the sound source; h is the vertical height of the sound source from the central point of the quaternary microphone array;
the phase difference between the array element A and the array element C is recorded as the phase difference between the first two array elements;
the phase difference between the B array element and the D array element is recorded as the phase difference between the second two array elements.
6. The method according to claim 5, wherein in step 3, the exact values of the first two-array element phase difference and the second two-array element phase difference calculated by the improved adaptive notch filter are realized by:
estimating the deviation degree of the signal frequency and the quantization frequency point, and carrying out frequency shift on the sound source signal acquired by the quaternary microphone array according to the deviation degree; the signal frequency is the maximum frequency in the frequency domain signals obtained after fast Fourier transform is carried out on the sound source signals, the quantization frequency point is the actual signal frequency obtained, and band-pass filtering is carried out on non-signal frequencies around the frequency domain signals obtained by the sound source signals to obtain noise-reduction sound source signals without other interference frequencies;
carrying out frequency domain filtering on the frequency-shifted signal, and carrying out inverse fast Fourier transform on the processed frequency domain filtering signal to reconstruct a time domain signal;
and performing correlation calculation according to the time domain signals after noise reduction to obtain the phase deviation of the received signals of each microphone, wherein the phase deviations are obtained by performing cross-correlation operation on the same array element, performing self-adaptive updating on the first coefficient and the second coefficient by using a self-adaptive algorithm, and performing position coordinate calculation to obtain the exact values of the phase difference between the first two array elements and the phase difference between the second two array elements.
7. The improved adaptive notch filter based indoor sound source localization method according to claim 1, wherein the input signal of the improved adaptive notch filter is represented as:
where x (k) is the received signal value formed by different discrete point values of the received signal, AxTo receive the amplitude of the signal, wk is the frequency of the input signal,
the signal phase deviation value of the receiving array element relative to the reference array element is obtained; n iskK is a discrete value of an acquired signal, and if m points are acquired in total to form an input signal, k represents from 1 to m;
from the cosine function properties:
after the input signal is noise-filtered, removing the noise value nkObtaining:
the output signal of the improved adaptive notch filter is represented as:
y(k)＝ws(k)rs(k)+wc(k)rc(k) (5)
in the formula, rs(k)、rc(k) Respectively sine reference signal, cosine reference signal, ws(k)、wc(k) The weight coefficients of the sine reference signal and the cosine reference signal are respectively, and omega represents the central frequency of the reference signal; r iss(k)＝Axsinwk、rc(k)＝Axcoswk、
The error formula of the improved adaptive filter is as follows:
e(k)＝x(k)-y(k) (6)
updating the weight coefficient of the sine reference signal and the weight coefficient of the cosine reference signal according to the feedback error value e (k):
ws(k+1)＝ws(k)+μe(k)rs(k) (7)
wc(k+1)＝wc(k)+μe(k)rc(k) (8)
where μ is the adaptive step size, e (k) is the error offset between the input signal and the iterated output result, ws(k +1) is the weight coefficient of the updated sinusoidal reference signal, wc(k +1) is the weight coefficient of the cosine reference signal after updating,
the phase estimation value in the actual operation process is obtained;
estimate the phase
Substituted into formula (4)
And obtaining a new fitting signal, then continuously iterating to obtain a phase estimation value until obtaining the phase offset caused by the actual sound source propagation delay, and when the error between the array received signal and the replica signal is smaller than a preset range, taking the correspondingly obtained phase estimation value as the phase offset caused by the actual delay to finally obtain the delay time.
8. The improved adaptive notch filter based indoor sound source localization method according to claim 7, wherein the adaptive step size is determined by the following formula:
wherein f issFor receiving the sampling frequency of the acoustic signal emitted by the source, BfFor full bandwidth, AxIs the amplitude of the input signal.
9. The method for indoor sound source localization according to claim 7, wherein the difference between the output signals y (k) of the two improved adaptive notches is calculated by taking the difference between the output signals y (k) of the two improved adaptive notches, and the calculation is performed by using the following formula:
in the formula (I), the compound is shown in the specification,
representing the phase difference between two array elements including the first two-array element phase difference
Second two-array element phase difference
Representing the phase of a microphone element,
Representing the phase of microphone element two; w is as1Weight coefficients, w, representing sinusoidal reference signals derived from a microphone element and a received signal ss2Weight coefficient, w, representing a sinusoidal reference signal derived on the basis of a microphone element number two and a received signal sc1Weight coefficient, w, representing a cosine reference signal derived on the basis of a microphone element and a received signal sc2Representing the weighting coefficients of a cosine reference signal derived on the basis of the microphone element number two and the received signal s.
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