CN111044973A - MVDR target sound source directional pickup method for microphone matrix - Google Patents
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- G—PHYSICS
- G01—MEASURING; TESTING
- G01S—RADIO DIRECTION-FINDING; RADIO NAVIGATION; DETERMINING DISTANCE OR VELOCITY BY USE OF RADIO WAVES; LOCATING OR PRESENCE-DETECTING BY USE OF THE REFLECTION OR RERADIATION OF RADIO WAVES; ANALOGOUS ARRANGEMENTS USING OTHER WAVES
- G01S5/00—Position-fixing by co-ordinating two or more direction or position line determinations; Position-fixing by co-ordinating two or more distance determinations
- G01S5/18—Position-fixing by co-ordinating two or more direction or position line determinations; Position-fixing by co-ordinating two or more distance determinations using ultrasonic, sonic, or infrasonic waves
- G01S5/20—Position of source determined by a plurality of spaced direction-finders
Abstract
The invention relates to the field of acoustic signal processing, and aims to provide a MVDR target sound source directional sound pickup method for a microphone matrix. The method has the advantages of small operation amount and high positioning precision, and can realize the directional acquisition of the multi-target sound source. The method comprises the following steps: collecting all sound source signals within a preset sound level range to obtain a sound source signal observation matrix; secondly, filtering, framing and the like are carried out on the sound source observation matrix, and a short-time spectrum is calculated; thirdly, determining the approximate azimuth of the target sound source corresponding to the peak position with the minimum delay on the cross-correlation curve by using a TDOA method; fourthly, determining the accurate position of the target sound source by utilizing an MVDR method in the approximate azimuth range of the target sound source; directionally collecting target sound source signals according to the accurate position of the target sound source; and sixthly, when two or more target sound sources exist, repeating the steps from three to five at the rest peak positions of the original cross-correlation curve until all the target sound sources are directionally picked up. The invention solves the practical requirement of multi-target sound signal acquisition.
Description
Technical Field
The invention relates to the technical field of acoustic signal processing, in particular to an MVDR target sound source directional sound pickup method for a microphone matrix.
Background
Microphone Array (Microphone Array) refers to an arrangement of microphones, that is to say a system consisting of a certain number of acoustic sensors (generally microphones) for sampling and processing the spatial characteristics of a sound field. Each microphone in the microphone array is called an array element.
Microphone arrays play a very important role in the processing of speech signals, with many advantages that cannot be achieved with a single microphone: firstly, the microphone array has a wider pickup range; secondly, the occurrence of the microphone array enables the processing of the voice signals not to be limited to a time domain and a frequency domain, and the voice signals can be processed according to the space coordinates, so that the further development of a sound source positioning algorithm is promoted; thirdly, the microphone array can collect more voice signals and perform data analysis on the voice signals, and more information about sound sources or information sources can be obtained. Due to the advantages of the microphone array, the microphone array can be widely used in voice signal processing technologies, such as large conference sites, video conferences, intelligent security auxiliary monitoring, robot hearing systems, and the like, and certainly, smart phones commonly used in life are also included.
The microphone array collects sound in all directions, various sounds are mixed together, and when some sound or some sounds need to be extracted, the sound needs to be acquired by using a sound source positioning algorithm. Currently, common sound source localization algorithms include Time difference of Arrival (TDOA) estimation, Direction of Arrival (DOA) estimation, and controlled beamforming (MVDR) techniques.
The TDOA estimation method is characterized in that time delay from a sound source to each array element of an array is estimated and calculated by an algorithm, and then the sound source position is obtained through a positioning algorithm. Although the DOA estimation method has good accuracy, it has a large amount of calculation, and is difficult to implement in a system. The MVDR can also be called a peak Power Response (SRP) based DOA estimation method, and the basic principle is to maximize the Response of the total output of the array to an incident signal in a certain specified direction through certain processing (such as weighting, delaying, and summing), and the method comprehensively considers the received information of each array element, so that the method has good anti-reverberation and anti-noise performance, but the estimation method is adopted to perform 360 ° scanning positioning, and the data computation amount is large, and is difficult to implement in system application.
Therefore, the current sound source positioning algorithm has respective defects and cannot meet the increasingly improved sound source positioning requirement.
Disclosure of Invention
Therefore, the technical problem to be solved by the present invention is that the sound source localization algorithms in the prior art have respective defects, and cannot meet the increasingly improved sound source localization requirements, so as to provide a directional sound pickup method for an MVDR target sound source of a microphone matrix, which has relatively small computation amount and high localization accuracy.
In order to solve the technical problems, the technical scheme adopted by the invention is as follows:
an MVDR target sound source directional pickup method for a microphone square matrix, the microphone square matrix comprising four array elements arranged in a square shape, the target sound source directional pickup method comprising the steps of:
collecting sound source signals of all sound intensity levels of the array elements within a preset sound intensity level range to obtain a sound source signal observation matrix received by the microphone array;
step two, filtering, normalizing, pre-emphasizing, denoising, windowing and framing the sound source signal observation matrix and calculating a short-time spectrum;
determining the approximate azimuth of a target sound source corresponding to the peak position with the minimum delay on the cross-correlation curve by using a TDOA estimation method;
fourthly, determining the accurate position of the target sound source by utilizing an MVDR estimation method in the approximate azimuth range of the target sound source;
directionally acquiring a target sound source signal by the microphone array according to the accurate position of the target sound source;
and step six, when the number of the target sound sources to be collected is two or more, repeating the steps three to five at the positions of the rest peak values of the original cross-correlation curve until the directional sound pickup of all the target sound sources is completed.
Preferably, the first step includes:
(1) acquiring the sound source signals of all the array elements with the sound intensity levels within the preset sound intensity level range, and taking one of the array elements as a reference array element, wherein the distance vector expression from the mth array element to the reference array element is rmThen the m-th array element receives the sound source signal xm(t) is expressed as:
wherein s (t) is the sound source signal, and k is the wave number vector;
(2) finding near-field steering vectors
① calculating the distance vector d from each sound source signal to each array elementms
Where Ψ is the incident angle of each sound source signal to the reference array element, d is the distance between two adjacent array elements, m represents the subscript of the array element, and r represents the subscript of the array elementsFor distance vectors, r, of individual sound source signals to reference array elementsmThe distance vector from the m array element to the reference array element is obtained;
② calculating the near field steering vector anear
Wherein k is a wave number vector, rsFor distance vectors of each acoustic source signal to a reference array element [ ·]TRepresents transposition;
(3) combining the sound source signals obtained by the four array elements to obtain a sound source signal observation matrix received by the microphone array;
from xm(t)，anearCalculating to obtain the popular matrix of the microphone array
Anear(θ)＝[anear(θ1)anear(θ2)…anear(θI)]
And the sound source signal observation matrix received by the microphone array
Xnear(t)＝Anear(θ)S(t)+N(t)。
Preferably, the wave number vector k is determined by a wave speed v and a frequency f, wherein the wave speed v is a sound speed, and v is 331.5+0.6TcM/s, TcIs the temperature of the drying air.
Preferably, the step of calculating the short-time spectrum in the second step is to use XnearAnd (t) dividing the microphone signals into frames according to time, and performing Fourier transform on the microphone signals in each frame to obtain the short-time spectrum of each microphone signal.
Preferably, the third step includes:
(1) calculating cross power spectrum of the sound source signal observation matrix
By using
Calculating the cross-power spectrum of the acoustic source signal observation matrix, where Ψ12(w) is a PHAT weighting function;
(2) carrying out weighting and Fourier inverse transformation processing on the cross-power spectrum to obtain a function peak value;
(3) obtaining the time delay estimation of each array element according to the position of the function peak value;
(4) and fitting a hyperbolic curve according to the time delay estimation of every two array elements, and calculating a common intersection point to obtain the approximate azimuth of the target sound source corresponding to the peak position with the minimum delay on the cross-correlation curve.
Preferably, the fourth step includes: computing beam responses
Scanning theta and psi places with high precision within the approximate azimuth range of the target sound sourceIt is possible to take the value that P (theta, psi) will be at the actual sound source orientation (theta)0,Ψ0) Taking a maximum value, thereby obtaining an accurate position (theta) of the target sound source0,Ψ0) (ii) a Wherein R isxxThe statistical covariance matrix of the array observation signals is replaced by a sample covariance matrix R of the sampled data,
where N is the number of sampling points obtained when the signal is sampled.
Preferably, the step five comprises:
(1) determining a weighted vector of a target sound source
According to the accurate position (theta) obtained in the fourth step0,Ψ0) Determining a weight vector according to the following formula
(2) And multiplying the sound source signals of the array elements by respective weighting vectors to obtain directional pickup data at the accurate position of the target sound source.
Preferably, the first step is preceded by the step of setting the preset sound intensity level range according to an application scenario.
Compared with the prior art, the technical scheme of the invention has the following advantages:
(1) the MVDR target sound source directional pickup method for the microphone array provided by the invention aims at the design of a square microphone array, firstly adopts a TDOA estimation method to carry out positioning estimation on a target sound source to obtain the approximate azimuth range of the target sound source, then utilizes the MVDR estimation method to carry out fine scanning in the approximate azimuth range of the target sound source, and takes the position with the strongest wave beam response in the approximate azimuth range of the target sound source as the accurate position of the target sound source. The TDOA estimation method and the MVDR estimation method are combined for use, so that the advantages of the two estimation methods are complemented, and the accuracy of sound source positioning is greatly improved on the premise of not increasing too much calculation amount.
(2) The MVDR target sound source directional pickup method for the microphone matrix provided by the invention can distinguish each target sound source when the target sound source is two or more, realizes directional collection of multi-target sound sources, can well improve the receiving effect and efficiency of the microphone by combining with an automatic control system, and can provide more diversified and high-quality application services for users by combining with other related technologies.
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In order that the present disclosure may be more readily and clearly understood, reference is now made to the following detailed description of the embodiments of the present disclosure taken in conjunction with the accompanying drawings, in which
FIG. 1 is a flow chart of the MVDR target sound source directional pickup method for a microphone matrix of the present invention;
FIG. 2 is a diagram of a near-field model to which the present invention relates;
FIG. 3 is a hyperbolic positioning map of the TDOA estimation method of the present invention.
Detailed Description
The technical solutions in the embodiments of the present invention will be clearly and completely described below with reference to the drawings in the embodiments of the present invention, and it is obvious that the described embodiments are only a part of the embodiments of the present invention, and not all of the embodiments. All other embodiments, which can be derived by a person skilled in the art from the embodiments given herein without making any creative effort, shall fall within the protection scope of the present invention.
Fig. 1 shows a preferred embodiment of the MVDR target sound source directional sound pickup method for microphone matrix according to the present invention. In the invention, the microphone square array comprises four array elements, each array element is a microphone, and the four microphones form a square area array. Directional sound pickup can be achieved using four omnidirectional microphones.
The invention relates to a MVDR target sound source directional pickup method for a microphone matrix, which specifically comprises the following steps:
collecting sound source signals of all sound intensity levels of each array element within a preset sound intensity level range to obtain a sound source signal observation matrix received by a microphone square matrix
(1) Acquiring sound source signals of all the sound intensity levels of all the array elements within a preset sound intensity level range
And before the first step, setting a preset sound intensity level range according to an application scene to prepare for subsequent sound source signal screening. Then, one of the array elements is used as a reference array element, and the distance vector from the m-th array element to the reference array element is expressed as rmThen the m-th array element receives the sound source signal xm(t) is expressed as:
where s (t) is the sound source signal and k is the wavenumber vector. The wave number vector k is jointly determined by a wave velocity v and a frequency f, wherein the wave velocity v is a sound velocity, and the wave velocity v is 331+0.6TcM/s, TcIs the temperature of the drying air.
(2) Finding near-field steering vectors
A near-field model is introduced to simulate the propagation process of a sound source signal, and a near-field guide vector is obtained, wherein the near-field model is shown in figure 2. The angle of incidence of the sound source signal cannot be seen as constant, and for each array element, the angle of incidence of different sound source signals is different.
① calculating the distance vector d from each sound source signal to each array elementms
Where Ψ is the incident angle of each sound source signal to the reference array element, d is the distance between two adjacent array elements, m represents the subscript of the array element, and r represents the subscript of the array elementsFor distance vectors, r, of individual sound source signals to reference array elementsmIs the distance vector from the m-th array element to the reference array element.
② calculating the near field steering vector anear
Wherein k is a wave number vector, rsFor distance vectors of each acoustic source signal to a reference array element [ ·]TRepresenting a transpose.
(3) Combining the sound source signals obtained by the four array elements to obtain a sound source signal observation matrix received by the microphone square matrix
From xm(t)，anearCalculating to obtain the popular matrix of the microphone array
Anear(θ)＝[anear(θ1)anear(θ2)…anear(θI)]
And the sound source signal observation matrix received by the microphone matrix
Xnear(t)＝Anear(θ)S(t)+N(t)。
Step two, filtering, normalizing, pre-emphasizing, de-noising, windowing and framing the sound source signal observation matrix and calculating a short-time spectrum
The method can remove noise and reduce the influence of the noise on subsequent sound source signal processing and acquisition. The step of calculating the short-time spectrum is to calculate XnearAnd (t) dividing the microphone signals into frames according to time, and performing Fourier transform on the microphone signals in each frame to obtain the short-time spectrum of each microphone signal. For two microphone signals, a cross power spectrum is calculated.
Step three, determining the approximate azimuth of the target sound source corresponding to the peak position with the minimum delay on the cross-correlation curve by using a TDOA estimation method
(1) Calculating the mutual power of the sound source signal observation matrix
By using
Cross-power spectrum of observation matrix of sound source signal is obtained, where psi12And (w) is a PHAT weighting function. By adopting the PHAT weighting function, the peak value is sharp when the signal-to-noise ratio is higher, and the anti-interference capability is stronger when the signal-to-noise ratio is lower.
(2) Weighting and Fourier inversion processing are carried out on the cross power spectrum to obtain the peak value of the function
(3) Obtaining the time delay estimation of each array element according to the position of the function peak value
(4) Fitting a hyperbolic curve according to the time delay estimation of every two array elements, calculating a common intersection point, and obtaining the approximate azimuth of the target sound source corresponding to the peak position with the minimum delay on the cross-correlation curve
As shown in fig. 3, a three-dimensional space positioning method is used to obtain the distance difference between a target sound source and each array element according to the time delay estimation of a signal, and the approximate azimuth range of the target sound source is obtained by fitting each hyperbola by using all two array elements as focuses, using the middle point of the two array elements as the origin of coordinates, using the distance obtained by the corresponding time delay estimation calculation as the distance difference, and calculating the common intersection point of each hyperbola.
Fourthly, determining the accurate position of the target sound source by utilizing an MVDR estimation method in the approximate azimuth range of the target sound source
And accurately scanning the estimated position by adopting an MVDR estimation method, taking the most prominent beam response azimuth in the approximate azimuth range of the target sound source as the accurate position of the target sound source, and then carrying out directional sound pickup. The MVDR is a single subband beam forming algorithm, and the target sound source signal is enhanced and noise and interference signals are suppressed by performing two-way focusing and weighting processing on the sound source signals received by the array, so that the directional output of the target sound source signal is free from distortion while the minimum output power is ensured.
In the fourth step, the beam response needs to be calculated
In the approximate azimuth range of the target sound source, all possible values of theta and psi are scanned with high precision, and P (theta, psi) is in the actual sound source azimuth (theta, psi)0,Ψ0) Taking a maximum value, thereby obtaining an accurate position (theta) of the target sound source0,Ψ0). Wherein R isxxThe statistical covariance matrix of the array observation signals is adopted, and in practical application, the sample covariance matrix of the sampling data is adoptedThe variance matrix R is replaced by a variance matrix R,
where N is the number of sampling points obtained when the signal is sampled.
Step five, directionally collecting the target sound source signals by the microphone square matrix according to the accurate position of the target sound source
(1) Determining a weighted vector of a target sound source
Although the gain of each array element in different directions is the same, when the sound source signals collected by different array elements are multiplied by different weighting coefficients, the directional diagram of the whole array points to the direction of the target sound source.
According to the accurate position (theta) obtained in the fourth step0,Ψ0) Determining a weight vector according to the following formula
(2) And multiplying the sound source signals of the array elements by the respective weighted vectors to obtain the directional sound pickup data at the accurate position of the target sound source.
Step six, when the number of the target sound sources to be collected is two or more, repeating the steps three to five on the positions of other peak values on the original cross-correlation curve until the directional sound pickup of all the target sound sources is completed
The MVDR target sound source directional pickup method for the microphone array is used for selectively screening various mixed sounds collected by the microphone elements by combining the TDOA estimation method and the MVDR estimation method, and directionally picking up sounds through spatial filtering, so that a target sound source is extracted. When the target sound sources are two or more, all the target sound sources can be separated, directional acquisition and pickup are realized, the positioning precision is high, and various application scenes can be met.
The following is detailed by combining specific embodiments of application scenarios:
in this embodiment, the application scenario is a video conference system, and the purpose of directional sound pickup is to avoid echo howling.
Firstly, the sound intensity level of the array collected signals is set within a specific range, and the too strong sound and the too weak sound are ignored. After the conference system is started, the relative direction between the microphone array and the loudspeaker is determined by utilizing the prompt sound emitted by the loudspeaker so as to exclude the direction in the subsequent orientation. When the array picks up the sound in the preset sound level range, the sound source signal observation matrix is circularly processed in batch through filtering, normalization, pre-emphasis, de-noising, windowing, framing and the like, then the approximate position of each speaker is determined by using the TDOA estimation result, then the accurate position of a target sound source is determined by using MVDR in the range, and finally the weighted vector is used for directionally collecting the sound of each speaker and transmitting the sound to a remote end.
It should be understood that the above examples are only for clarity of illustration and are not intended to limit the embodiments. Other variations and modifications will be apparent to persons skilled in the art in light of the above description. And are neither required nor exhaustive of all embodiments. And obvious variations or modifications therefrom are within the scope of the invention.
Claims (8)
1. An MVDR target sound source directional sound pickup method for a microphone square matrix, wherein the microphone square matrix comprises four array elements which are arranged in a square shape, and the target sound source directional sound pickup method is characterized by comprising the following steps of:
collecting sound source signals of all sound intensity levels of the array elements within a preset sound intensity level range to obtain a sound source signal observation matrix received by the microphone array;
step two, filtering, normalizing, pre-emphasizing, denoising, windowing and framing the sound source signal observation matrix and calculating a short-time spectrum;
determining the approximate azimuth of a target sound source corresponding to the peak position with the minimum delay on the cross-correlation curve by using a TDOA estimation method;
fourthly, determining the accurate position of the target sound source by utilizing an MVDR estimation method in the approximate azimuth range of the target sound source;
directionally acquiring a target sound source signal by the microphone array according to the accurate position of the target sound source;
and step six, when the number of the target sound sources to be collected is two or more, repeating the steps three to five at the positions of the rest peak values of the original cross-correlation curve until the directional sound pickup of all the target sound sources is completed.
2. The MVDR target sound source directional sound pickup method for microphone arrays as claimed in claim 1, wherein the step one comprises:
(1) acquiring the sound source signals of all the array elements with the sound intensity levels within the preset sound intensity level range, and taking one of the array elements as a reference array element, wherein the distance vector expression from the mth array element to the reference array element is rmThen the m-th array element receives the sound source signal xm(t) is expressed as:
wherein s (t) is the sound source signal, and k is the wave number vector;
(2) finding near-field steering vectors
① calculating the distance vector d from each sound source signal to each array elementms
Where Ψ is the incident angle of each sound source signal to the reference array element, d is the distance between two adjacent array elements, m represents the subscript of the array element, and r represents the subscript of the array elementsFor distance vectors, r, of individual sound source signals to reference array elementsmThe distance vector from the m array element to the reference array element is obtained;
② calculating near field steeringVector anear
Wherein k is a wave number vector, rsFor distance vectors of each acoustic source signal to a reference array element [ ·]TRepresents transposition;
(3) combining the sound source signals obtained by the four array elements to obtain a sound source signal observation matrix received by the microphone array;
from xm(t)，anearCalculating to obtain the popular matrix of the microphone array
Anear(θ)＝[anear(θ1)anear(θ2)...anear(θI)]
And the sound source signal observation matrix received by the microphone array
Xnear(t)＝Anear(θ)S(t)+N(t)。
3. The MVDR target sound source directional sound pickup method for a microphone matrix of claim 2, wherein: the wave number vector k is jointly determined by a wave velocity v and a frequency f, wherein the wave velocity v is a sound velocity, and v is 331.5+0.6TcM/s, TcIs the temperature of the drying air.
4. The MVDR target sound source directional sound pickup method for microphone matrix according to claim 2 or 3, characterized in that: the step of calculating the short-time spectrum in the step two is to use XnearAnd (t) dividing the microphone signals into frames according to time, and performing Fourier transform on the microphone signals in each frame to obtain the short-time spectrum of each microphone signal.
5. The MVDR target sound source directional sound pickup method for microphone arrays as claimed in claim 4, wherein said step three comprises:
(1) calculating cross power spectrum of the sound source signal observation matrix
By using
Calculating the cross-power spectrum of the acoustic source signal observation matrix, where Ψ12(w) is a PHAT weighting function;
(2) carrying out weighting and Fourier inverse transformation processing on the cross-power spectrum to obtain a function peak value;
(3) obtaining the time delay estimation of each array element according to the position of the function peak value;
(4) and fitting a hyperbolic curve according to the time delay estimation of every two array elements, and calculating a common intersection point to obtain the approximate azimuth of the target sound source corresponding to the peak position with the minimum delay on the cross-correlation curve.
6. The MVDR target sound source directional sound pickup method for microphone arrays as claimed in claim 5, wherein said step four comprises: computing beam responses
In the approximate azimuth range of the target sound source, scanning all possible values of theta and psi with high precision, and P (theta, psi) is to be at the actual sound source azimuth (theta, psi)0，Ψ0) Taking a maximum value, thereby obtaining an accurate position (theta) of the target sound source0，Ψ0) (ii) a Wherein R isxxThe statistical covariance matrix of the array observation signals is replaced by a sample covariance matrix R of the sampled data,
where N is the number of sampling points obtained when the signal is sampled.
7. The MVDR target sound source directional sound pickup method for microphone matrix according to claim 6, wherein said step five comprises:
(1) determining a weighted vector of a target sound source
According to the accurate position (theta) obtained in the fourth step0，Ψ0) Determining a weight vector according to the following formula
(2) And multiplying the sound source signals of the array elements by respective weighting vectors to obtain directional pickup data at the accurate position of the target sound source.
8. The MVDR target sound source directional sound pickup method for a microphone matrix as claimed in claim 1, wherein: the first step is preceded by the step of setting the preset sound level range according to an application scene.
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