CN102831898A - Microphone array voice enhancement device with sound source direction tracking function and method thereof - Google Patents
Microphone array voice enhancement device with sound source direction tracking function and method thereof Download PDFInfo
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Abstract
The invention provides a microphone array voice enhancement device with a sound source direction tracking function and a method for the microphone array voice enhancement device. The microphone array voice enhancement device relates to voice signal processing. A microphone array, an adjustable parallel beam former group module, a fixed parameter FIR (Finite Impulse Response) filter module, a fixed parameter signal blocking module, an adaptive noise canceller module and a sound source direction update module are aranged in the devide. The method comprises the following steps of: initializing, forming an adjustable beam, fixing parameter filtering, blocking a signal, carrying out adaptive noise cancelling, and updating a sound source direction. The invention provides an adjustable parallel beam former group combined with a sidelobe canceller structure to achieve real-time tracking of a target sound source direction, embeds the sound source direction tracking function directly to a generalized sidelobe canceller structure, can achieve sound source direction tracking and voice enhancement, thereby overcoming sensibility of an algorithm performance on a DOA (Direction of Arrival) estimation error.
Description
Technical field
The present invention relates to a kind of voice signal and handle, especially relate to a kind of microphone array speech sound enhancement device and method thereof with the Sounnd source direction following function.
Background technology
A frontier that is the current speech signal Processing based on the research and the application of microphone array.The influence that the voice signal that microphone receives in field of voice signal such as speech recognition, voice control, phonetic synthesis receives neighbourhood noise and interference is very big; Had a strong impact on the processing quality of voice signal, the general speech-enhancement system based on single microphone is difficult to obtain reinforced effects preferably.Microphone array has been owing to utilized the spatial information of echo signal, noise and interference, based on the speech-enhancement system of microphone array better reinforced effects can be provided.
Array microphone; Can a plurality of microphones be formed an array according to the topological structure of design; The signal that collects like this increases a spatial domain again on the basis of time-frequency domain, can carry out space and time diversity to the multipath signal that collects and handle, and microphone array can form different responses to the signal on the different directions; It also is the spatial direction characteristic of array; Make array microphone have auditory localization and tracking, voice extraction and separate and function such as denoising, thereby improve the quality of speech signal under complex background, remedy and isolate the defective that microphone could obtain and utilize spatial information.
Nineteen eighty-two Griffiths and Jim (1, L.J.Griffiths; C.W.Jim.An alternative approach to linearly constrained adaptive beamforming.IEEE Transactions on Antennas and Propagation.January; 1982,30,27-34) propose to revise linear Beam-former; It is generalized side lobe canceller (Generalized Sidelobe Canceller is called for short GSC).Generalized side lobe canceller allows the response of Beam-former is controlled widely; Especially can be converted into non-limited self-adaptation to the limited problem of linear restriction finds the solution; Thereby have more general meaning, in microphone array voice enhancement process, obtained extensive studies and application.To the deficiency of classical GSC algorithm in practical application; People such as Gannot (2, Sharon Gannot; Israel Cohen.Speech Enhancement Based on the General TransferFunction GSC and Postfiltering.IEEE Transactions on Speech and Audio Processing.2004; 12 (6)) be the basis with classical generalized sidelobe canceller algorithm, proposed a kind of based on the non-stationary acoustics transfer function generalized side lobe canceller of useful signal.People such as Abad (3.A Abad; J Hernando.Speech Enhancement and recognition by Integrating Adaptive Beamforming and Wiener Filtering.IEEE Sensor Array and Multichannel Signal Processing Workshop; SAM; Sitges, 2004) the non-self-adapting branch road that then proposes Wiener filtering is introduced GSC further improves the effect that secondary lobe is eliminated.
It is to be noted; The accurate direction of obtaining sound source is to utilize the generalized side lobe canceller technology to carry out the prerequisite of microphone array voice enhancement process, has only obtained behind the target sound source orientation to be listed in the expectation Sounnd source direction through adaptive algorithm training microphone array and to form wave beam and realize that voice strengthen minimizing under the output variance criterion.
Chinese patent ZL 200510105526.7 discloses a kind of multi-channel adaptive speech signal disposal route of using noise to reduce, and this method increases the signal to noise ratio (S/N ratio) that an adaptive processor improves signalling channel through the fixed beam path to GSC.But this method still need estimate to compensate each channel time delay by the frequency domain time delay, so that wave beam is to the quasiexpectation Sounnd source direction.
But classical generalized sidelobe canceller algorithm remains unchanged to the wave beam of quasiexpectation passage in the voice enhanced processes all the time, the expection sense of having only after location algorithm recomputates target direction just once more set algorithm to aim at.In the actual application that microphone array voice such as video conference, speech recognition, Speaker Identification strengthen, often have the occasion that the speaker moves in the process of speaking.At this moment, from the angle that the microphone array voice strengthen, require the sensing wave beam of microphone array can aim at mobile target speaker all the time to obtain best voice reinforced effects.And according to traditional generalized sidelobe canceller algorithm structure; Need carry out the auditory localization computing earlier; And then carry out GSC microphone array voice enhancement process according to the target sound source orientation that obtains; Can cause having hysteresis between expection sound bearing that GSC aims at and actual sound bearing, thereby influence the voice reinforced effects.
Simultaneously; Because the classical generalized sidelobe canceller need obtain the target sound source direction and move precondition as algorithm; Sounnd source direction (DOA when obtaining target sound source direction and actual Sounnd source direction certain error is arranged; Direction ofarrival) mismatch can have influence on the blockage effect of blocking matrix to the Sounnd source direction wanted signal; Be the noise cancellation device input end that the part signal of non-self-adapting branch road leaks into the self-adaptation branch road, cause voice signal to be weakened, thereby have influence on the voice reinforced effects.
Summary of the invention
First purpose of the present invention is to provide a kind of microphone array speech sound enhancement device with the Sounnd source direction following function.
Second purpose of the present invention is to provide a kind of microphone array voice enhancement method that adopts the microphone array speech sound enhancement device of said band Sounnd source direction following function.
The microphone array speech sound enhancement device of said band Sounnd source direction following function is provided with:
Microphone array is used for voice signal multichannel collecting, pre-process and analog to digital conversion;
Adjustable parallel Beam-former pack module is used for through adjusting the real-time follow-up that each channel time delay carries out Sounnd source direction;
Fixed coefficient FIR filter module is used for forming desired frequency characteristics at the non-self-adapting branch road and obtains signals and associated noises;
Fixed coefficient signal jam module is used for the signal that the filtering Sounnd source direction comprised and obtains noise signal;
Adaptive noise canceller module is used for offseting the noise that principle is eliminated signals and associated noises as a reference and comprised with noise signal according to adaptive noise, output voice enhancing signal;
The Sounnd source direction update module; Be used for carrying out the voice enhancing signal variance selection current optimal Sounnd source direction that secondary lobe is eliminated output, and the current optimal Sounnd source direction that obtains is inputed to the offset direction, the left and right sides that adjustable parallel Beam-former pack module upgrades current Sounnd source direction and current Sounnd source direction according to current expection Sounnd source direction and offset direction, current expection sound source left and right sides correspondence;
Each passage voice signal output terminal behind pre-amplification circuit and analog to digital converter, directly is connected with the signal input part of adjustable parallel Beam-former pack module through data line successively in the said microphone array;
Said adjustable parallel Beam-former pack module is provided with 3 parallel adjustable Beam-formers; Said 3 parallel adjustable Beam-formers are respectively equipped with signal input part and Sounnd source direction input end; Each signal input part connects each passage voice signal output terminal of analog to digital converter, and each Sounnd source direction input end then is connected with the current optimal Sounnd source direction output terminal of Sounnd source direction update module; The input end through adjusted each the passage voice signal output termination fixed coefficient signal jam module of time delay of said 3 parallel adjustable Beam-formers, the stack output of each passage voice signal of time delay adjustment back then is connected with the input end of fixed coefficient FIR filter module;
Said Adaptive Noise Canceler module is provided with reference noise input end and noisy speech input end; Fixed coefficient signal jam module is blocked the reference noise input end that output after the processing connects the Adaptive Noise Canceler module to input signal, and fixed coefficient FIR filter module carries out the noisy speech input end that output after the Filtering Processing connects the Adaptive Noise Canceler module to input signal;
The voice enhancing signal output termination Sounnd source direction update module of said Adaptive Noise Canceler module is to carry out the selection of current optimal Sounnd source direction; The Adaptive Noise Canceler module is provided with current optimal Sounnd source direction output terminal and voice enhancing signal output terminal; Current optimal Sounnd source direction output termination adjustable parallel Beam-former pack module, the corresponding voice enhancing signal of voice enhancing signal output terminal output current optimal Sounnd source direction.
Said microphone array can adopt the equidistant linear array of being made up of 5 yuan of microphones.
Said microphone array voice enhancement method adopts the microphone array speech sound enhancement device of said band Sounnd source direction following function, said method comprising the steps of:
1 initialization step: at initial phase the dead ahead is set and is the current audio direction of acquiescence, or import the Sounnd source direction that obtains by the microphone array location algorithm, as the initialization Sounnd source direction, as the Sounnd source direction parameter of adjustable Beam-former pack module;
1 adjustable wave beam forms step: according to the Sounnd source direction adjustment step-length of setting; Adding, subtract the adjustment step-length respectively through current Sounnd source direction produces the left and right offset direction of current Sounnd source direction and calculates the corresponding delay compensation value of each channel signal of microphone array; And to 3 parallel beams form each passage output signal of microphone array in the device carry out the counterparty to delay compensation, make it aim at Sounnd source direction and offset direction, the left and right sides respectively;
1 fixed coefficient filter step: utilize fixed coefficient respectively FIR filtering to be carried out in the output of 3 parallel beams formation devices, be used for forming the frequency response of needs at current Sounnd source direction, 3 the wave beam aligning directions in offset direction, the left and right sides;
1 signal jam step: at the self-adaptation branch road, the signal of 3 adjustable Beam-former outputs is imported 3 identical signal jam matrixes respectively, is used for the voice signal that comprises in the current Sounnd source direction of filtering, 3 the wave beam aligning directions in offset direction, the left and right sides;
1 adaptive noise removal process: at the self-adaptation branch road; With the corresponding blocking matrix output signal of 3 wave beams of current Sounnd source direction, offset direction, left and right sides noise signal as a reference, fixedly the output signal of corresponding 3 wave beams carries out adaptive noise and eliminates processing as the band input signal of making an uproar in the branch road;
1 Sounnd source direction step of updating: output poor that obtains the output of 3 beam direction non-self-adapting tributary signals and corresponding Adaptive Noise Canceler; And select the minimum beam direction of variance to upgrade in the view window as current Sounnd source direction, upgrade and add, subtract the adjustment step-length behind the current Sounnd source direction respectively and produce the left and right offset direction of current Sounnd source direction and calculate the corresponding delay compensation value of each channel signal of microphone array and carry out algorithm iteration once more.
The problem that the present invention will solve is that a kind of microphone array speech sound enhancement device with the Sounnd source direction following function is provided on the basis of traditional generalized side lobe canceller algorithm.The situation that the speaker possibly be moved in strengthening to microphone array voice such as Speaker Identification, speech recognitions; The present invention provides a kind of adjustable parallel beam to form device group combination sidelobe cancellation structure and realizes the real-time follow-up to the target sound source direction; The Sounnd source direction following function is directly embedded the generalized side lobe canceller structure; Can realize that Sounnd source direction is followed the tracks of, voice carry out when strengthening, thereby can overcome the susceptibility of algorithm performance to the DOA evaluated error.
Technical scheme of the present invention is on the basis of traditional generalized side lobe canceller, to add the enhancement process that the Sounnd source direction following function is carried out voice signal.
The microphone array speech sound enhancement device of band Sounnd source direction following function provided by the invention realizes that the concrete thinking that band Sounnd source direction tracking microphone array voice strengthen is: at first the non-self-adapting signal branch in sidelobe canceller is provided with 3 parallel Beam-formers composition Beam-former groups; Each Beam-former is aimed at current Sounnd source direction successively, is reached the left and right sides direction of displacement of current Sounnd source direction (offset direction, the left and right sides obtains through a current Sounnd source direction direction adjustment of plus-minus step-length respectively); Calculate the microphone array wave beam and aim at the output of sidelobe canceller sef-adapting filter under above-mentioned 3 expectation Sounnd source direction conditions respectively; And in a view window, carry out performance relatively, select current best Sounnd source direction according to the comparative result that 3 Sounnd source directions are corresponding; Obtaining behind the current best Sounnd source direction it to be made as current Sounnd source direction, and obtaining new Sounnd source direction left and right sides drift direction, carrying out the direction iteration of parallel Beam-former group once more; Through above-mentioned iterative process, realize that the microphone array voice that the band Sounnd source direction is followed the tracks of strengthen.Therefore; The basic structure that band Sounnd source direction disclosed by the invention is followed the tracks of the enhancing of microphone array voice is actually 3 multiple operations, the adjustable sidelobe canceller of direction; Wherein each sidelobe canceller is distinguished the offset direction, the left and right sides of corresponding current Sounnd source direction and current Sounnd source direction; And select through the iteration optimization that the output signal variance of each sidelobe canceller is carried out current Sounnd source direction, in iterative process, carry out current Sounnd source direction simultaneously through adaptive algorithm and follow the tracks of and the voice enhancing.
Compare with sound enhancement method with existing microphone array location; The microphone array speech sound enhancement device of band of the present invention sound bearing following function has 3 outstanding advantages: first; Form the device group owing in the secondary lobe Canceller, embed adjustable parallel beam; Therefore can realize carrying out when tracking of acoustic target direction and microphone array voice strengthen, need not to rely on extra microphone array location algorithm and carry out the target sound source direction calculating; Second; Because embedded parallel beam forms the device group and in algorithm flow, strengthens operation simultaneously with voice all the time in the voice signal processing procedure; Therefore can guarantee carrying out when voice strengthen the target sound source direction being carried out real-time follow-up, thereby can realize quick, real-time response the moving target audio source tracking; The 3rd; When there are error in the Sounnd source direction that obtains and actual Sounnd source direction, carry out the sound bearing through adjustable parallel Beam-former group and follow the tracks of, can eliminate the mismatch of voice signal arrival direction (DOA); Thereby improve the signal leakage in the self-adaptive path, improve voice and strengthen the property.
Description of drawings
Fig. 1 is the microphone array speech sound enhancement device example structure composition frame chart of band Sounnd source direction following function according to the invention.
Fig. 2 be the embodiment of the invention 5 yuan of microphone arrays and with microprocessor CC figure.
Fig. 3 is that data stream, the control stream of each signal processing module in the embodiment of the invention connects synoptic diagram.
Fig. 4 is the adjustable Beam-former structural representation of the embodiment of the invention.
Fig. 5 is the schematic diagram calculation that the ultrasound wave auxiliary microphone voice of the embodiment of the invention strengthen each channel time delay offset.
Embodiment
In order to make technology contents of the present invention, characteristic, advantage more obviously understandable, following examples will combine accompanying drawing that the present invention is further described.
As shown in Figure 1, the microphone array speech sound enhancement device embodiment of said band Sounnd source direction following function is provided with microphone array 1, adjustable parallel Beam-former pack module 2, fixed coefficient FIR filter module 3, fixed coefficient signal jam module 4, adaptive noise canceller module 5 and Sounnd source direction update module 6.
Each passage voice signal output terminal behind pre-amplification circuit and analog to digital converter 7, directly is connected with the signal input part of adjustable parallel Beam-former pack module 2 through data line successively in the said microphone array 1.
Said adjustable parallel Beam-former pack module 2 is provided with 3 parallel adjustable Beam-formers; Said 3 parallel adjustable Beam-formers are respectively equipped with signal input part and Sounnd source direction input end; Each signal input part connects each passage voice signal output terminal of analog to digital converter, and each Sounnd source direction input end then is connected with the current optimal Sounnd source direction output terminal of Sounnd source direction update module 6; The input end through adjusted each the passage voice signal output termination fixed coefficient signal jam module 4 of time delay of said 3 parallel adjustable Beam-formers, the stack output of each passage voice signal of time delay adjustment back then is connected with the input end of fixed coefficient FIR filter module 3.
Said Adaptive Noise Canceler module 5 is provided with reference noise input end and noisy speech input end; 4 pairs of input signals of fixed coefficient signal jam module block the reference noise input end that output after the processing connects Adaptive Noise Canceler module 5, and 3 pairs of input signals of fixed coefficient FIR filter module carry out the noisy speech input end that output after the Filtering Processing connects Adaptive Noise Canceler module 5.
The voice enhancing signal output termination Sounnd source direction update module 6 of said Adaptive Noise Canceler module 5 is to carry out the selection of current optimal Sounnd source direction; Adaptive Noise Canceler module 5 is provided with current optimal Sounnd source direction output terminal and voice enhancing signal output terminal; Current optimal Sounnd source direction output termination adjustable parallel Beam-former pack module 2, the corresponding voice enhancing signal of voice enhancing signal output terminal output current optimal Sounnd source direction.
Said microphone array 1 can adopt the equidistant linear array of being made up of 5 yuan of microphones.
Microphone array is by 5 microphone (m0 that equidistantly arrange among the microphone array speech sound enhancement device embodiment of said band Sounnd source direction following function; M1; M4) form the microphone linear array, the voice signal that each microphone obtains in the array is sent into the sidelobe canceller of embedded 3 parallel adjustable Beam-former groups and is carried out Sounnd source direction tracking and voice enhancing.
Microphone array is made up of microphone and hardware circuit; The pressure type electret microphone mic0 that wherein microphone array is little, simple in structure by volume, electroacoustic performance is good; ...; Mic4, pre-amplification circuit that NJM2100 operational amplifier chip constitutes and MAX118 modulus conversion chip constitute (as shown in Figure 2), microphone space d=10cm in the present embodiment.
Composition modules such as adjustable parallel Beam-former pack module, fixed coefficient FIR filter module, fixed coefficient signal jam module, adaptive noise canceller module, Sounnd source direction update module all belong to digital signal processing module, adopt ARM9 S3C2440 microprocessor to carry out software programming in the present embodiment and realize.
The connected mode of microphone array and microprocessor is: 52 grades of pre-amplification circuits amplification back input hyperchannel modulus conversion chip MAX118 that microphone output signal constitutes through operational amplifier shown in Figure 2 in the microphone array; The S3C2440 microprocessor is through IO mouth GPB2; 3; Input channel end A1, A2, the A3 of 4 control MAX118; Through timer output pin TOUT0, TOUT1 control MAX118 read/write the analog to digital conversion that inbound port WR, RD carry out SF 16ksps, carry out the transmission of 8bit analog to digital conversion result through data line DATA0 to DATA7 to the S3C2440 microprocessor.
After the number conversion of multicenter voice signal mode gets into microprocessor in the microphone array speech sound enhancement device of said band Sounnd source direction following function; As shown in Figure 3 with the data between each digital signal processing module of form of software operation, control stream connected mode, specify as follows:
The adjustable Beam-former group that the microphone array speech sound enhancement device of said band Sounnd source direction following function is introduced is that 3 identical parallel beams of structure form device; And can adopt two kinds of methods to carry out the initialization of adjustable Beam-former pack module parameter in the device: the one, the dead ahead can be set be the current audio direction of acquiescence, algorithm begins can converge to real current Sounnd source direction gradually after the following function; The 2nd, the Sounnd source direction that microphone array location algorithm well known in the art capable of using obtains is as current Sounnd source direction, and algorithm begins after the following function current Sounnd source direction to be followed the tracks of.
Adjustable Beam-former pack module is after initial phase obtains current Sounnd source direction information; Be formed on the offset direction, the left and right sides of each shifted by delta about Sounnd source direction according to the Sounnd source direction adjustment step delta of setting; 3 parallel beams form devices and aim at Sounnd source direction and offset direction, the left and right sides respectively, in each parallel beam formation device, are target sound source direction or offset direction, the left and right sides each channel input signal x of corresponding adjustment microphone array according to its aligning promptly also
i(k) i=0,1,2 ..., 5 delay compensation value τ
i(θ
j) i=0,1,2 ..., 5, j=0,1,2, obtain each the channel signal x behind the delay compensation thus
I, j(k), wherein i represents each channel number of microphone array, θ
jRepresent the corresponding respectively current Sounnd source direction that points to of 3 adjustable Beam-formers, current sound source left avertence is moved, right avertence is moved direction.In the present embodiment, θ
1Be current Sounnd source direction, θ
0, θ
2Be respectively the left and right offset direction of current Sounnd source direction, Δ is a wave beam adjustment step-length.Promptly have:
θ
0＝θ
1-Δ,θ
2＝θ
1+Δ
Through behind delay compensation-additional wave beam shaping, three road microphone array wave beams that can be aimed at current Sounnd source direction, offset direction, the left and right sides respectively form signal y
j(k) j=0,1,2
y
j(k)＝A
TX
i,j(k)
In the present embodiment, adopt simple delay compensation-additional wave beam shaping, fixed coefficient A=[0.2,0.2,0.2,0.2,0.2].
3 parallel adjustable Beam-formers are as shown in Figure 4 in the adjustable parallel Beam-former pack module, respectively are provided with signal input, two input ends of Sounnd source direction input, and adjusted each the passage voice signal of time delay is exported, wave beam forms two output terminals of output.Each adjustable Beam-former can be used for forming the Sounnd source direction that the voice wave beam is aimed at input, and 3 parallel adjustable Beam-formers then can be aimed at current Sounnd source direction and offset direction, the left and right sides thereof respectively.
As shown in Figure 3; The wave beam of 3 parallel adjustable Beam-former outputs forms signal and imports 3 identical fixed coefficient FIR wave filters respectively; The coefficient of this FIR wave filter is calculated by the frequency response of Sounnd source direction expection and obtains, and is used for forming at current Sounnd source direction, 3 the wave beam aligning directions in offset direction, the left and right sides frequency response of needs.
y'
j(k)＝F
TY(k)
As shown in Figure 3 equally; Adjusted each passage voice signal of time delay of 3 parallel adjustable Beam-former outputs is imported 3 identical signal jam matrixes respectively, is used for the voice signal that comprises in the current Sounnd source direction of filtering, 3 the wave beam aligning directions in offset direction, the left and right sides.The signal jam matrix B is configured according to following formula in the present embodiment:
After the blocking matrix processing, 3 corresponding signal jams of beam direction are output as:
U
j(k)＝B
TX
i,j(k)
3 Adaptive Noise Canceler modules shown in Figure 3 are equipped with reference noise, two input ends of signals and associated noises; And export signal noise input signal as a reference with the blocking matrix of current Sounnd source direction, 3 the wave beams correspondences in offset direction, the left and right sides respectively; The output signal of corresponding 3 wave beams of fixed coefficient FIR wave filter adopts the weight coefficient W of LMS well known in the art (least-mean-square error algorithm) adaptive algorithm adjustment Adaptive Noise Canceler as noisy input signal in the present embodiment
J, kCarry out adaptive noise and eliminate processing.
At this moment, 3 corresponding Adaptive Noise Canceler of beam direction are output as:
n
j(k)＝W
j,k TU
j(k)
After noise removing was handled, 3 corresponding system voices of beam direction strengthened the poor of the output that is output as each wave beam non-self-adapting tributary signal output and Adaptive Noise Canceler:
s
j(k)＝y'
j(k)-n
j(k)
Obtain 3 corresponding system voices of beam direction and strengthened output s
j(k) after, direction update module shown in Figure 3 take the length L as the variance V of the view window length voice enhanced output signal that relatively three each beam directions are corresponding
jAnd select the minimum beam direction of variance to carry out algorithm and upgrade with the residual noise in the minimizing voice signal as new current Sounnd source direction; L gets 200 in the present embodiment, and promptly per 200 algorithm iterations calculate the cost function of once representing with the output signal variance and carry out once current Sounnd source direction renewal.
Simultaneously, can strengthen output s through the voice of best Sounnd source direction output this method in the direction update module
JAnd obtain new current Sounnd source direction θ (k),
1new=θ
J
As shown in Figure 3, obtaining new current Sounnd source direction θ
1newAfter, according to this Sounnd source direction and offset direction, left and right sides θ thereof
0new, θ
2newUpgrade each channel time delay offset τ
i(θ
Jnew) i=0,1,2 ..., 5, j=0,1,2, proceed algorithm iteration.In the algorithm iteration process, eliminate tracking and the voice enhancing that computing has realized Sounnd source direction simultaneously through 3 adaptive noises of aiming at beam direction.
In Fig. 3, adjustable parallel Beam-former pack module is made up of 3 parallel adjustable Beam-formers.Wherein the concrete structure of each adjustable Beam-former is as shown in Figure 4, and each adjustable Beam-former is equipped with signal input, two input ends of Sounnd source direction input, and adjusted each the passage voice signal of time delay is exported, wave beam forms two output terminals of output.After the speech data of passage 0 to 4 input was adjusted through each channel time delay, the one tunnel directly exported adjusted each channel data of time delay, and other one the tunnel carries out each passage stack obtains wave beam formation output.Wherein each passage of 3 parallel adjustable Beam-formers adjustment time delay respectively by the input Sounnd source direction, and input Sounnd source direction plus-minus wave beam adjustment step-length obtain.Therefore, each adjustable Beam-former can be used for forming the Sounnd source direction that the voice wave beam is aimed at input, and 3 parallel adjustable Beam-formers then can be aimed at current Sounnd source direction and offset direction, the left and right sides thereof respectively.
In the above-mentioned course of work, need be according to the Sounnd source direction θ that aims at
jCalculate each channel time delay offset of microphone array, this principle described below in conjunction with Fig. 5:
As shown in Figure 5, in embodiments of the present invention: with 5 yuan of microphone linear array place horizontal lines is the X axle, is that true origin is set up elements of a fix system with the microphone m2 position in the middle of the linear array, and each array element distance of linear array is d, is θ in azimuth of target then
jThe time; Consider that sound source a is in far-field range among the embodiment; When arriving the microphone linear array, its voice signal that sends to think the plane incident wave; Then the center array element microphone m2 with the present embodiment linear array carries out corresponding delay compensation value calculating as benchmark, and the voice signal that promptly m2 is received is not made delay compensation, to the voice signal x of each passage microphone reception in the linear array
iCan be according to azimuth angle theta
jCarry out following delay compensation and calculate (as shown in Figure 5):
x
i,j(k)＝x
i(k')
Wherein i is the numbering of each passage in the linear array, and C is the airborne velocity of sound (getting 340m/s in the present embodiment), θ
jThe corresponding respectively current Sounnd source direction that points to of three adjustable Beam-formers is represented in (j=0,1,2), and current sound source left avertence is moved, right avertence is moved direction, f
sBe the SF (unit is Hz, is 16000Hz in the present embodiment) of microphone array voice signal, round () represents rounding operation.Each passage voice signal carries out weighted stacking after through three corresponding delay compensations of wave beam, can realize aiming at the offset direction, the left and right sides of current Sounnd source direction, current sound.
The maximum characteristics of microphone array speech sound enhancement device that band Sounnd source direction disclosed by the invention is followed the tracks of and method thereof are can in the voice enhancement process, carry out the tracking of Sounnd source direction by 3 parallel adjustable Beam-former combining adaptive noise eliminators; There is certain error the sound bearing of importing when allowing initialization simultaneously; Owing to have the following function of Sounnd source direction, the signal leakage that can avoid sound bearing (DOA) mismatch to cause.
Claims (3)
1. be with the microphone array speech sound enhancement device of Sounnd source direction following function, it is characterized in that being provided with:
Microphone array is used for voice signal multichannel collecting, pre-process and analog to digital conversion;
Adjustable parallel Beam-former pack module is used for through adjusting the real-time follow-up that each channel time delay carries out Sounnd source direction;
Fixed coefficient FIR filter module is used for forming desired frequency characteristics at the non-self-adapting branch road and obtains signals and associated noises;
Fixed coefficient signal jam module is used for the signal that the filtering Sounnd source direction comprised and obtains noise signal;
Adaptive noise canceller module is used for offseting the noise that principle is eliminated signals and associated noises as a reference and comprised with noise signal according to adaptive noise, output voice enhancing signal;
The Sounnd source direction update module; Be used for carrying out the voice enhancing signal variance selection current optimal Sounnd source direction that secondary lobe is eliminated output, and the current optimal Sounnd source direction that obtains is inputed to the offset direction, the left and right sides that adjustable parallel Beam-former pack module upgrades current Sounnd source direction and current Sounnd source direction according to current expection Sounnd source direction and offset direction, current expection sound source left and right sides correspondence;
Each passage voice signal output terminal behind pre-amplification circuit and analog to digital converter, directly is connected with the signal input part of adjustable parallel Beam-former pack module through data line successively in the said microphone array;
Said adjustable parallel Beam-former pack module is provided with 3 parallel adjustable Beam-formers; Said 3 parallel adjustable Beam-formers are respectively equipped with signal input part and Sounnd source direction input end; Each signal input part connects each passage voice signal output terminal of analog to digital converter, and each Sounnd source direction input end then is connected with the current optimal Sounnd source direction output terminal of Sounnd source direction update module; The input end through adjusted each the passage voice signal output termination fixed coefficient signal jam module of time delay of said 3 parallel adjustable Beam-formers, the stack output of each passage voice signal of time delay adjustment back then is connected with the input end of fixed coefficient FIR filter module;
Said Adaptive Noise Canceler module is provided with reference noise input end and noisy speech input end; Fixed coefficient signal jam module is blocked the reference noise input end that output after the processing connects the Adaptive Noise Canceler module to input signal, and fixed coefficient FIR filter module carries out the noisy speech input end that output after the Filtering Processing connects the Adaptive Noise Canceler module to input signal;
The voice enhancing signal output termination Sounnd source direction update module of said Adaptive Noise Canceler module is to carry out the selection of current optimal Sounnd source direction; The Adaptive Noise Canceler module is provided with current optimal Sounnd source direction output terminal and voice enhancing signal output terminal; Current optimal Sounnd source direction output termination adjustable parallel Beam-former pack module, the corresponding voice enhancing signal of voice enhancing signal output terminal output current optimal Sounnd source direction.
2. the microphone array speech sound enhancement device of band Sounnd source direction following function as claimed in claim 1 is characterized in that said microphone array adopts the equidistant linear array of being made up of 5 yuan of microphones.
3. microphone array voice enhancement method is characterized in that adopting according to claim 1 with the microphone array speech sound enhancement device of Sounnd source direction following function, said method comprising the steps of:
1 initialization step: at initial phase the dead ahead is set and is the current audio direction of acquiescence, or import the Sounnd source direction that obtains by the microphone array location algorithm, as the initialization Sounnd source direction, as the Sounnd source direction parameter of adjustable Beam-former pack module;
1 adjustable wave beam forms step: according to the Sounnd source direction adjustment step-length of setting; Adding, subtract the adjustment step-length respectively through current Sounnd source direction produces the left and right offset direction of current Sounnd source direction and calculates the corresponding delay compensation value of each channel signal of microphone array; And to 3 parallel beams form each passage output signal of microphone array in the device carry out the counterparty to delay compensation, make it aim at Sounnd source direction and offset direction, the left and right sides respectively;
1 fixed coefficient filter step: utilize fixed coefficient respectively FIR filtering to be carried out in the output of 3 parallel beams formation devices, be used for forming the frequency response of needs at current Sounnd source direction, 3 the wave beam aligning directions in offset direction, the left and right sides;
1 signal jam step: at the self-adaptation branch road, the signal of 3 adjustable Beam-former outputs is imported 3 identical signal jam matrixes respectively, is used for the voice signal that comprises in the current Sounnd source direction of filtering, 3 the wave beam aligning directions in offset direction, the left and right sides;
1 adaptive noise removal process: at the self-adaptation branch road; With the corresponding blocking matrix output signal of 3 wave beams of current Sounnd source direction, offset direction, left and right sides noise signal as a reference, fixedly the output signal of corresponding 3 wave beams carries out adaptive noise and eliminates processing as the band input signal of making an uproar in the branch road;
1 Sounnd source direction step of updating: output poor that obtains the output of 3 beam direction non-self-adapting tributary signals and corresponding Adaptive Noise Canceler; And select the minimum beam direction of variance to upgrade in the view window as current Sounnd source direction, upgrade and add, subtract the adjustment step-length behind the current Sounnd source direction respectively and produce the left and right offset direction of current Sounnd source direction and calculate the corresponding delay compensation value of each channel signal of microphone array and carry out algorithm iteration once more.
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