CN1967658A - Small scale microphone array speech enhancement system and method - Google Patents
Small scale microphone array speech enhancement system and method Download PDFInfo
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- CN1967658A CN1967658A CNA2005100868763A CN200510086876A CN1967658A CN 1967658 A CN1967658 A CN 1967658A CN A2005100868763 A CNA2005100868763 A CN A2005100868763A CN 200510086876 A CN200510086876 A CN 200510086876A CN 1967658 A CN1967658 A CN 1967658A
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Abstract
The invention provides the small-scale microphone array speech enhancement system and method. The system in the invention consists of two-part, signal acquisition and signal processing. Signal acquisition part comprises the microphone array and the AD converter component. Noisy voice signal is collected from the microphone as analog signals, through the AD converter, turning into digital signals, getting signal enhancement in the DSP chip, and then through USB port, transmitting to the computer for preservation. Signal processing part comprises the microphone consistency calibration module, the adaptive noise cancellation module, the SNR (Signal Noise Ratio) estimation module, and acoustic source location adaptive module, etc. The system in the invention can remove the introduced environmental noise in collection process, increase the collected acoustic source SNR, and obtain better acoustic source signal. In most cases the sound source signal is the speech signal, speech intelligibility improves a little after de-noising process.
Description
Technical field
The invention belongs to the signal processing technology field, be specifically related to a kind of anti-noise system and method, comprise anti-noise system and correlation technique based on minitype microphone array.
Background technology
Along with the development that mechanics of communication and Digital Signal Processing make rapid progress, people are more and more higher to the requirement of multimedia communication, and technology such as car handsfree conversation, video conference are used widely more and more.In these are used, though vision signal can strengthen the quantity of information of communication, make that interchange is more natural, the interchange between the people mainly still relies on language to finish, if the quality of sound does not guarantee that it is meaningless that communication also will become.So the quality of audio signal quality has determined the performance of whole communication system, whether voice are clear understandable, are the basic indexs of differentiating these multimedia systems.
Yet, the ground unrest in the reality, the sound that RMR room reverb and other interference source send can greatly destroy the quality of voice signal.Traditional frequency domain noise-reduction method carries out filtering to signal according to the feature of signal spectrum and noise spectrum on frequency domain, the voice signal of stationary noise signal and non-stationary is separated.But, if noise signal also right and wrong stably, traditional noise-reduction method has run into difficulty.As in noisy cocktail party, other people have just become noise by one's voice in speech, and this and main speaker's the statistical property of voice on frequency domain is very similar, and only starting with from frequency domain, to isolate useful voice signal very difficult.But the speech-enhancement system of using microphone array then can make the receiving gain of total system spatially form directive property by certain methods, just spatially selects interested signal to strengthen, and promptly carries out so-called " spatial filtering ".
In recent years, used the method for Array Signal Processing and carried out the development of voice technique for enhancing rapidly, more and more be applied in the systems such as speech communication, man-machine interaction.
Using the array signal method carries out signal and strengthens problem to be solved and be: in the acoustic enviroment of complexity, improve the receiving gain of whole array on specific direction by regulating each path filter coefficient, reduce the receiving gain of undesired signal on other directions simultaneously, thereby improve the signal to noise ratio (S/N ratio) of output signal.
Based on the problems referred to above, a lot of people have carried out the research of this respect.Face following two problems at present:
1. can only the sound-source signal of fixed-direction be strengthened, so just make enhanced system that significant limitation has been arranged.When sound source position changes, must make the directive property of array aim at sound source position by the angle of adjusting array.But in a lot of occasions, this artificial adjustment is unusual inconvenience, as long-range video conference etc.
2. the scale problem of array.Microphone array yardstick in the past is big (being generally more than the 50cm), and this makes and carry inconvenience, can't use on mobile device.
If the problems referred to above are resolved, then microphone array speech enhancement system just can obtain following application:
1. gather the front end of voice signal as osophone, improve the intelligibility of acquired signal;
2. be used in the ambulatory handheld voice capture device, such as mobile phone, intercom, hand-held sound-track engraving apparatus etc.;
3. in vehicle-mounted interactive voice control system, the utilization microphone array can make driver's some parts action by the password controls car, and as the switch vehicle window, the angular adjustment of reflective mirror etc. greatly facilitate the driver;
4. when being used in the interview equipment, need not the position of mobile collection system, just can in certain distance, follow the tracks of, make the voice collecting effect improve by interviewer's signal.
Summary of the invention
Purpose of the present invention is exactly the problems referred to above that solve microphone array, a kind of microphone array speech enhancement system and method based on small scale proposed, can remove the neighbourhood noise of introducing in the gatherer process, make the sound-source signal signal to noise ratio (S/N ratio) that collects improve, thereby obtain better sound-source signal.Sound-source signal is a voice signal under most situation, increases through the intelligibility of denoising to voice.
Small scale microphone array speech enhancement system of the present invention mainly is divided into signals collecting and two subdivisions of signal Processing.
The signals collecting part is made up of microphone array and AD converter.Noisy speech signal is a simulating signal by the microphone collection, changes into digital signal by AD converter, carries out signal and strengthen in the DSP process chip, and then be passed on the computing machine by the usb oral instructions and preserve.
Microphone array is made up of 3 omni-directional microphone that become equilateral triangle to place, and is responsible for gathering the voice signal of simulation.So-called omnidirectional is exactly that the receiving gain of microphone on all directions of space is all identical.Put like this and can obtain 6 groups of different directivity indexs, thereby when the sound source position adaptation module obtains sound source position, can select suitable directivity index according to this locality by adaptive approach.
AD converter then is responsible for this simulating signal is converted into digital signal, handles for DSP.
Signal processing is by microphone consistance correction module, adaptive noise cancellation module, compositions such as SNR estimation module and sound source position adaptation module.
Microphone consistance correction module is made up of three FIR wave filters, one of them wave filter is an all-pass filter, effect makes that just the signal of three passages is identical through the time-delay after this module, two other wave filter is by measuring the transport function of three microphones, forms based on the transfer function configures of three microphones;
The adaptive noise cancellation module is made up of adaptive noise elimination wave filter, and wherein structure is based on improved self-adaptation zero notch bundle formation method; The exponent number of sef-adapting filter can be set according to the noise intensity that the SNR estimation module obtains.
The SNR estimation module is the energy by calculating noise, and by comparing with the threshold value of setting, output is the control signal to adaptive noise cancellation module median filter length then.
The sound source position adaptation module is by Beam-former, form with the speech energy analyzer, by constructing tropism's the wave beam of not remaking, analyze the speech energy in each wave beam then, obtain the approximate location of source of students, send control signal then, make the directivity index of adaptive noise cancellation module point to Sounnd source direction.
On the other hand, small scale microphone array sound enhancement method of the present invention mainly is divided into two big steps: signals collecting and signal Processing.
Noisy speech signal is a simulating signal by the microphone collection, changes into digital signal by AD converter, carries out signal and strengthen in the DSP process chip, and then be passed on the computing machine by the usb oral instructions and preserve.
Signal Processing is then by steps such as the correction of microphone consistance, adaptive noise elimination, SNR estimation and sound source position self-adaptations.
The microphone consistance is proofreaied and correct: this step is at first by playing calibrating signal, adopt the relative transport function between two microphones of lowest mean square (LMS) method acquisition, make the signal that collects by this function then, remove the influence that microphone transfer function is accepted signal, thereby effectively proofreaied and correct the inconsistency between the microphone, make Enhancement Method insensitive, improved the robustness of Enhancement Method the difference of microphone.
Adaptive noise is eliminated: the self-adaptation zero notch bundle formation module by the variable filter exponent number can be constructed forward direction heart and the directive property of back to heart, the noise signal that obtains comprising the voice signal of noise and do not contain voice, noise signal by comprising in the sef-adapting filter filtering voice signal obtains pure voice signal then.
SNR estimation: under the condition of low signal-to-noise ratio, will make multistage sef-adapting filter be difficult to convergence.This module is carried out guestimate by the signal to noise ratio (S/N ratio) to signal, and the exponent number with wave filter under the situation of low signal-to-noise ratio reduces, and makes wave filter can successfully restrain under any state of signal-to-noise, and this step has been guaranteed the accuracy that adaptive noise is eliminated.
Sound source position self-adaptive processing: the received signal when at first obtaining different directions gain maximum by the directive property of regulating microphone array, then phonetic element is wherein analyzed, find the approximate location of sound source, and this positional information passed to the adaptive noise cancellation module, adjust the directivity index of array, make signal strengthen direction and aim at sound source position.
Main characteristics of the present invention is:
1. Du Te three microphone speech enhancement technique are more desirable more than dual microphone enhancing effect in the past;
2. the Null-Forming method is improved, used the sef-adapting filter that changes length, the system that makes obtains better to strengthen effect;
3. add microphone consistance alignment technique, although make with seldom number of microphone, still can well be strengthened effect.This is achieved with regard to the microphone array that makes small scale;
4. by the different microphone directive property of structure, gather the signal on the different directions, analytic signal characteristic is then found out the position of target sound source, then the directive property of array is adjusted, and reaches the purpose of following the tracks of sound-source signal;
5. come the length of real-time regulated sef-adapting filter by detection, can well remove the noise of non-stationary to noise energy.
Description of drawings
Below in conjunction with accompanying drawing the present invention is illustrated in further detail:
Fig. 1 is a signal acquisition part subsystem framework synoptic diagram;
Fig. 2 is a microphone array shape synoptic diagram;
Fig. 3 is signal processing flow figure;
Fig. 4 is improved self-adaptation zero a notch bundle formation method synoptic diagram;
Fig. 5 is a microphone array directive property synoptic diagram.
Embodiment
Below with reference to accompanying drawing of the present invention, most preferred embodiment of the present invention is described in more detail.
Small scale microphone array speech enhancement system of the present invention mainly is divided into signals collecting and two subdivisions of signal Processing.
Figure 1 shows that signal acquisition part subsystem framework synoptic diagram, the signals collecting part is made up of microphone array and AD converter.Noisy speech signal is a simulating signal by the microphone collection, changes into digital signal by AD converter, carries out signal and strengthen in the DSP process chip, and then be passed on the computing machine by the usb oral instructions and preserve.
1. signals collecting is partly implemented principle and method
Native system carries out signals collecting by the microphone of 3 omnidirectionals to sound source, and microphone array becomes equilateral triangle, and the leg-of-mutton length of side is 2.54cm, as shown in Figure 2
The simulating signal that microphone array collects is converted into 16kHz sampling, the digital signal that 16bit quantizes through the AD chip of Ti company. carry out signal Processing by the multiple signals disposable plates then.Wherein, the microphone consistance is proofreaied and correct, and adaptive noise is eliminated, and modules such as SNR estimation and sound source position self-adaptation all realize by the TI 5509DSP chip on the plate.USB interface by system board is transferred to that computer is play or data are preserved at last.
2. the enforcement principle and the method for signal processing
The signal processing method flow is: gather three road signals by three microphones, two-way is wherein used improved self-adaptation zero notch bundle formation method, carrying out adaptive noise eliminates, carry out the estimation of noise energy simultaneously, estimated result is passed to the adaptive noise cancellation module, regulate the length of wave filter.
By three microphone configurations difference microphone arrays,, obtain the signal of all directions simultaneously, analyze the sound source position that draws roughly, pass to the adaptive noise cancellation module then, select to strengthen direction by changing directive property.Figure 3 shows that the method flow of signal processing
1) microphone consistance correction module
Owing to manufacturing process and and other many-sided reasons, the frequency response of two microphones and delay character all have some difference, the error that this may cause method result of calculation makes the method performance descend.In order to adapt to various microphone, the universality of raising method, this paper is by measuring the correlativity of two microphone input signals, investigate the difference between two passages, utilize adaptive approach, obtain being calibrated the shock response curve of passage with respect to reference channel, thereby make two microphones with sound source equidistant the time, the signal of output is identical.
Because the frequency response curve of microphone can be very not flat in the reality, can not think an all-pass filter.So establish H1 here, H2 is their shock response separately.Add a wave filter F in the H2 back among the present invention, make the output signal minimum.
If input signal is s (t), the transport function of microphone is respectively h
1(t), h
2(t).
If the compensating filter transport function c (t) that obtains at last then has
s(t)*h
1(t)＝s(t)*h
2(t)*c(t)
h
1(t)＝h
2(t)*c(t)
In order to try to achieve c (t), we do not remove directly to ask h
2 -1But adopt the method for auto adapted filtering (t).As shown in the figure, make the minimum power of the difference e (t) of two passages output, promptly
Min|e(t)|
2＝(s
out1(t)-s
out2(t)*c(t))
2
Can obtain c (t) coefficient by lowest mean square (LMS) method.Obtain in 3 microphones 2 relatively and the transport function c of another microphone by said method
1(t), c
2(t).The input of these two microphones is passed through with c
1(t), c
2(t) be the wave filter of coefficient, the another one microphone is through length and c
1(t) identical all-pass filter, the purpose of making is in order not introduce additional time delay like this.
2) noise energy estimation module
In improved Nullforming method, the back is a noise reference signal to the resulting signal y of heart-shaped directive property (n), this module is by calculating the energy of y (n), control the length of sef-adapting filter, divide frame to handle signal, 410 sampling points are a frame, the normalized energy of getting 10 frames is as criterion, and carry out Noise Estimation one time every 1s, doing one side like this is for fear of the length owing to frequent change wave filter, make the convergence of wave filter be affected, also saved calculated amount on the other hand.
If the maximal value of AD conversion is A, then have
If E (n)＞500, the exponent number of then adjusting sef-adapting filter is 1.Reason is that noise is very big at this moment, and the wave filter VAD on long rank estimates inaccurate, and it is bad to cause restraining effect, needs to adopt the zero limit of self-adaptation wave filter.
If 200＜E (n)＜500, the exponent number of then adjusting sef-adapting filter is 12.
If E (n)＜200, the exponent number of then adjusting sef-adapting filter is 16.
3) sound source position adaptation module
By the utilization difference method, regulate the delay inequality after two microphone signals subtract each other, can control the direction at microphone maximum gain orientation and zero point.
Gain formula by the input of binary channels microphone difference
H(θ)＝τ+d/ccosθ
By regulating the directive property of τ acquisition different directions, the present invention obtains 8 groups of different in space directivity indexs by changing the τ value, energy ingredient to the signal analysis voice that obtain occupy therein detects by VAD, thinks that the direction of speech signal energy maximum is the direction of sound-source signal.This module passes to the adaptive noise cancellation module with this information, the direction that control strengthens.
4) improved self-adaptation zero notch bundle forms (Null-Forming)
Native system adopts improved Null-Forming method, and this is a kind of adaptive voice Enhancement Method based on difference microphone array technology.In schematic diagram as shown in Figure 4, the signal that front end microphone (front microphone) and rear end microphone (back microphone) are received is respectively f (n) and b (n), d is the spacing of two microphones, corresponding d/c is the time delay of two passages, the coefficient of W (n) sef-adapting filter, a (n) is the output of sef-adapting filter, and z (n) is the output of total system.
The signal of b (n) subtracts each other with front end f (n) through after postponing d/c.The directive property of x (n) is the heart of forward direction, and zero point is at 180 °.On the contrary, the directive property of y (n) be the back to heart, zero point is at 0 °.In this method, we think that y (n) is a noise reference signal, and x (n) is noisy sound-source signal.
The polar coordinates figure that the output z (n) of total system is corresponding is obtained through sef-adapting filter W (n) by x (n) and y (n).Different with traditional Null-Forming method, the sef-adapting filter W (n) that the present invention adopts is not simple open loop form single order LMS wave filter, but sef-adapting filter that exponent number is variable, according to the output exponent number of Noise Estimation module on 1 rank, change between 12 rank and 16 rank, thereby make wave filter can in the noise of non-stationary, obtain good effect.
Under the very noisy situation, W (n) is a simple open loop form single order LMS wave filter.Because do not need sound end to detect, can very fast reaction be arranged to noise; Under the small noise situation, W (n) is a closed loop high-order LMS wave filter, and the directive property of this wave filter is stronger, and better anti-reverberation ability is also arranged.
When this module is received sound source position information, select two in three microphones, by improved Null-Forming method, make to strengthen the direction that direction is aimed at sound source.And three microphones are equilateral triangle and distribute, and can obtain (A-F) 6 groups of different enhancing directions according to the characteristics of Null-Forming, as shown in Figure 5.So just the Null-Forming method than dual microphone has had more flexibility.
Although disclose specific embodiments of the invention and accompanying drawing for the purpose of illustration, its purpose is to help to understand content of the present invention and implement according to this, but it will be appreciated by those skilled in the art that: without departing from the spirit and scope of the invention and the appended claims, various replacements, variation and modification all are possible.Therefore, the present invention should not be limited to most preferred embodiment and the disclosed content of accompanying drawing.
Claims (8)
1. a small scale microphone array speech enhancement system comprises two subdivisions of signals collecting and signal Processing, and concrete structure is as follows:
1) the signals collecting part is made up of microphone array and AD converter, and microphone array is made up of 3 omni-directional microphone that become equilateral triangle to place;
2) signal processing is made up of microphone consistance correction module, adaptive noise cancellation module, SNR estimation module and sound source position adaptation module:
Microphone consistance correction module is made up of three FIR wave filters, and one of them wave filter is an all-pass filter, and two other wave filter forms based on the transfer function configures of three microphones;
The adaptive noise cancellation module is eliminated wave filter by adaptive noise and is formed, and it forms method construct based on improved self-adaptation zero notch bundle and forms;
The SNR estimation module is used for the energy of calculating noise, and the threshold value of result of calculation and setting is compared, and output is to the control signal of adaptive noise cancellation module median filter length;
The sound source position adaptation module is made up of Beam-former and speech energy analyzer.
2. small scale microphone array speech enhancement system as claimed in claim 1 is characterized in that: described omnidirectional is meant that the receiving gain of microphone on all directions of space is all identical.
3. small scale microphone array speech enhancement system as claimed in claim 1 is characterized in that: the exponent number of sef-adapting filter is set according to the noise intensity that the SNR estimation module obtains.
4. a small scale microphone array sound enhancement method comprises signals collecting and signal Processing two big steps, and concrete steps are as follows:
Signals collecting is to be simulating signal by microphone with the noisy speech signal collection, changes into digital signal by AD converter, carries out signal and strengthen in the DSP process chip, is transferred on the computing machine then and preserves;
Signal Processing comprises that the signal to gathering carries out: the microphone consistance is proofreaied and correct, and adaptive noise is eliminated, SNR estimation, sound source position self-adaptive processing.
5. small scale microphone array sound enhancement method as claimed in claim 4, it is characterized in that, the microphone consistance is proofreaied and correct the method that adopts: at first by playing calibrating signal, adopt the relative transport function between two microphones of least-square methods acquisition, make the signal that collects by this function then, remove the influence that microphone transfer function is accepted signal.
6. small scale microphone array sound enhancement method as claimed in claim 4, it is characterized in that, the method of adaptive noise elimination employing is: the self-adaptation zero notch bundle by the variable filter exponent number forms heart of structure forward direction and the directive property of back to heart, the noise signal that obtains comprising the voice signal of noise and do not contain voice, noise signal by comprising in the sef-adapting filter filtering voice signal obtains pure voice signal then.
7. small scale microphone array sound enhancement method as claimed in claim 4, it is characterized in that, the method that SNR estimation adopts is: the exponent number with wave filter under the situation of low signal-to-noise ratio reduces, and wave filter can successfully be restrained under any state of signal-to-noise.
8. small scale microphone array sound enhancement method as claimed in claim 4, it is characterized in that, the method of sound source position self-adaptive processing employing is: the received signal when at first obtaining different directions gain maximum by the directive property of regulating microphone array, then phonetic element is wherein analyzed, find the position range of sound source, and adjust the directivity index of array according to this positional information, make signal strengthen direction and aim at sound source position.
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