CN1295678C - Subband adaptive valley point noise reduction system and method - Google Patents
Subband adaptive valley point noise reduction system and method Download PDFInfo
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Abstract
本发明公开一种子带自适应谷点降噪系统和方法，传声器对拾取的向前和向后指向性信号输入分析滤波器，分解为多个向前和向后子带信号，在自适应滤波器上将相应频带的向后子带信号乘以自适应滤波系数，输出向后子带调整信号，再在加法器上将各频带的向前子带信号减去向后子带调整信号输出，最后由综合滤波器组将多个加法器的输出累加，输出处理完的信号。本发明通过在频率上分带，在几个噪声源分布在不同的频率段并在不同方位的情况下，不同频率段的谷点可以同时对准几个噪声源，确保输出信号中噪声得到抑制，当采用单指向性传声器对时，对传声器的要求低，效果稳定，可以在相当小的空间尺寸内实现。
The invention discloses a sub-band self-adaptive valley point noise reduction system and method. A microphone inputs the picked-up forward and backward directional signals to an analysis filter, decomposes them into a plurality of forward and backward sub-band signals, and performs adaptive filtering The backward sub-band signal of the corresponding frequency band is multiplied by the adaptive filter coefficient on the device, and the backward sub-band adjustment signal is output, and then the forward sub-band signal of each frequency band is subtracted from the backward sub-band adjustment signal output on the adder, and finally The output of multiple adders is accumulated by the integrated filter bank, and the processed signal is output. In the present invention, by dividing bands on frequency, when several noise sources are distributed in different frequency bands and in different orientations, the valley points of different frequency bands can be aligned with several noise sources at the same time, so as to ensure that the noise in the output signal is suppressed , when a unidirectional microphone pair is used, the requirements for the microphone are low, the effect is stable, and it can be realized in a relatively small space size.
Description
Technical field
The present invention relates to a kind of voice de-noising method, more particularly, the present invention relates to a kind of voice de-noising method of adaptive pointing.
Background technology
There is the hearing device (or instrument) of directive property to utilize echo signal source and noise signal source to realize the enhancing of echo signal and the inhibition of noise signal usually from the characteristics of different directions.At present this directive acoustical signal pickup apparatus has two kinds of patterns: a kind of have a fixed directional, comprise that single directional microphone and two microphones are to (dipole) etc., so-called fixed directional just is meant at fixed-direction and strengthens and decay, can not change along with the orientation of target sound source and noise source and change; Another kind of its directive property is adaptive, can come the Adjustment System parameter according to the variation in the orientation of target sound source and noise source, with the enhancing that dynamically realizes echo signal and the inhibition of noise signal, be that directive property is variable, the environment that becomes and move when noise source can reach effect preferably.
At Fa-Long Luo, Jun Yang, Senior Member, Chaslav Pavlovic, and AryeNehorai, " the Adaptive Null-Forming Scheme in Digital Hearing Aids " that is shown, IEEE Trans.Signal Processing, vol.50, no.7, a kind of existing adaptive pointing microphone system is disclosed among the JULY 2002., please refer to the system chart shown in Fig. 1, what adopt is that the non-directive microphone is right, and the signal that a passage (front microphone) and two passages (backmicrophone) are received is respectively f (n) and b (n), and corresponding d/c is the time delay of two passages, x (n) and y (n) are forward and the backward directive property signals that calculates, W is the coefficient of each band-adaptive wave filter, and a (n) is the output of sef-adapting filter, and z (n) is the output of total system.
This system can come the Adjustment System parameter according to the variation in the orientation of target sound source and noise source, with the enhancing that dynamically realizes echo signal and the inhibition of noise signal.But during some is used, such as osophone, mobile phone, PDA, meeting room Mike etc., their environment of applications are normally Protean, noise source is very complicated, they may be in different directions, and be distributed in the different frequency section, for example, at meeting room, the fan noise of projector is arranged, it almost is a kind of single-frequency noise, traffic noise from is outside window arranged, they are some low-frequency noises, and these two kinds of noises make just to be difficult to simultaneously relatively good with these two kinds of noise removing in this way from different directions for Mike.
In addition, this system requires height to the manufacturing accuracy of microphone, and two microphones need have higher output consistance, and removing the effect of making an uproar can bigger reduction take place along with the microphone changes of properties, and bulk is bigger in addition.
Summary of the invention
The technical problem to be solved in the present invention provides a kind of sub-band adaptive valley point noise-reduction method, can effectively eliminate to be distributed in the noise that different frequency section and a plurality of noise sources on different azimuth are sent.The present invention also will provide a kind of system that realizes this method.
In order to solve the problems of the technologies described above, the invention provides a kind of sub-band adaptive valley point noise-reduction method, may further comprise the steps:
(a) the microphone group is picked up directive property signal and directive property signal forward backward respectively, and the signal energy that system receives in different directions is variant, and the direction of received energy minimum is exactly the valley point;
(b) be decomposed into a plurality of subband signals forward and a plurality of subband signal backward with directive property signal backward in identical frequency band division mode forward with described;
(c) to each frequency band, with subband signal forward and backward the cross correlation function of subband signal obtain the adaptive filter coefficient of this frequency band divided by the power of subband signal backward;
(d) on each frequency band, deduct the product of subband signal and this band-adaptive filter coefficient backward with subband signal forward, again with all difference additions, the voice signal that obtains handling.
Said method can have following characteristics: forward described and backward the directive property signal be digital sampled signal.
A kind of sub-band adaptive provided by the invention valley point noise reduction system comprises microphone group, two analysis filterbank, a plurality of sef-adapting filter, a plurality of totalizer and synthesis filter groups, wherein:
Described microphone group is used for pick-up of acoustic signals, produces forward and directive property signal backward, and the signal energy that system receives in different directions is variant, and the direction of institute's received energy minimum is exactly the valley point;
Described two analysis filterbank are respectively applied for forward, the directive property signal is decomposed into a plurality of subband signals forward and a plurality of subband signal backward by identical frequency band division mode backward;
Described a plurality of sef-adapting filter, the subband signal backward that is respectively applied for frequency band multiply by the auto adapted filtering coefficient, export backward subband and adjust signal, described auto adapted filtering coefficient equal this frequency band forward subband signal and backward the cross correlation function of subband signal divided by the power of subband signal backward;
Described a plurality of totalizer, the subband signal forward that is respectively applied for each frequency band deducts subband adjustment signal and output backward;
Described synthesis filter group is used for the output of all totalizers being added up the signal that obtains handling.
Said system can have following characteristics: comprise sampling and AD conversion unit in the described microphone group, the analog acoustic signal that is used for picking up is converted to digital sampled signal.
Said system can have following characteristics: described microphone group is that unidirectional microphines is right.
As from the foregoing, the present invention is distributed under different frequency bands and the situation in different azimuth in several noise sources by divide band on frequency, and the valley point of different frequency section can be aimed at several noise sources simultaneously, guarantees that noise is inhibited in the output signal.In addition, adopt unidirectional microphines to the time, the manufacturing accuracy of microphone is required low, effect stability can be realized in quite little bulk, range of application enlarges.
Description of drawings
Fig. 1 is the system chart of existing adaptive pointing microphone;
Fig. 2 is the synoptic diagram that unidirectional microphines receives the directive property of 1kHz acoustical signal;
Fig. 3 is the synoptic diagram that unidirectional microphines is used sub-band adaptive valley point noise reduction system;
Fig. 4 is the functional block diagram of the right sub-band adaptive valley point noise reduction system of embodiment of the invention unidirectional microphines.
Embodiment
When unidirectional microphines received the 1kHz acoustical signal, in the response of all directions as shown in Figure 2, thick line was represented the normalized energy of the signal that array receives on all directions.It is variant to have the signal energy that directive property that is to say that system receives in different directions.The valley point is exactly the direction of received energy minimum, i.e. the degree of 180 among figure direction.
When target sound source and noise source are in different azimuth, as shown in Figure 3, unidirectional microphines is opposite between the two, and one of them microphone definite object sound source is defined as microphone forward, the signal that picks up is x (n), be called directive property signal forward, another microphone direction is opposite with it, is defined as microphone backward, the signal that picks up is y (n), is called directive property signal backward.If target sound source is positioned at 0 degree direction, then the polar coordinates figure of the directive property correspondence of x (n) is the heart shape diagram of valley point at 180 degree, and the corresponding polar coordinates figure of same y (n) is the heart shape diagram of valley point at 0 degree.In addition, comprise in the microphone of present embodiment the sampling and AD conversion unit, forward described and backward the directive property signal be digital sampled signal.
Fig. 4 is the functional block diagram of embodiment of the invention system, as shown in the figure, signal x (n) and y (n) send into analysis filterbank 11 respectively and carry out frequency division, obtain a plurality of subband signals forward and subband signal x (n backward, f) and y (n, f), again with each subband signal y (n backward, f) send into the sef-adapting filter 12 of this frequency band respectively, W
iIt is the coefficient of each band-adaptive wave filter, a (n, f) be that sef-adapting filter is exported subband adjustment signal backward, the x of subband signal forward (n with each frequency band, f) and backward subband is adjusted signal a (n, f) after each totalizer is subtracted each other, comprehensively just obtained the output signal z (n) of system again through synthesis filter group 13.
Valley point and the coefficient W (n) of the output z of system (n) have following relation:
Wherein, f is the frequency of signal, θ
NttllIt is the angle value of valley point correspondence.
Formula (1.1) has shown valley point and adaptation coefficient W, and (n, relation f) is to the dependence of signal frequency.Carry out the branch band by the signal that x (n) and these two of y (n) is had fixed directional by frequency domain, in each frequency band, carry out Adaptive Identification respectively, obtain adaptive filter coefficient W (n separately, f), (n f) changes the time-delay of each frequency band, just can change the polar coordinates figure of output correspondence to regulate W, make the valley point of total system aim at noise source, realize reducing the purpose of noise.
To specify adaptation coefficient W (n below, f) computing method, suppose that s (n) and i (n) are useful signal (voice signal) and the noise signals that enters in the signal of two passages, 0 ° of direction is aimed in the valley point of microphone backward, voice signal can obtain following various on ° direction of θ=0:
x(n)＝s(n)+i(n) (1.2)
y(n)＝i(n) (1.3)
a(n，f)＝W(n，f)y(n，f) (1.4)
Promptly directive property signal y (n) only contains noise component backward, and in present specification, the directive property signal is meant the valley point sound source that aims at the mark backward, only contains the pickoff signals of noise component.These characteristics show that reducing system's output power is equivalent to the power that reduces the noise component in the output power, because voice signal and noise signal are uncorrelated.The expectation of the power of the output z of system (n) is as follows:
R wherein
Xx(n, f) and R
Yy(n, f) be respectively x (n, f) and y (n, power f), R
Xy(n, f) be x (n, f) and y (n, cross correlation function f).Formula (1.6) is that (n, quadratic expression f) has and only have a minimum value W about W
Opt, making its minimum, can obtain:
According to above analysis, (n f), just can guarantee output signal z (n, f) direction of subtend noise source in the valley point of each frequency band to utilize formula (1.7) to calculate the coefficient W of each sef-adapting filter in the present embodiment system.
Accordingly, present embodiment self-adaptation valley point noise-reduction method may further comprise the steps:
Step 110 is picked up directive property signal and directive property signal forward backward respectively;
Step 130, to each frequency band, with subband signal x forward (n, f) and backward (n, cross correlation function f) obtain the adaptive filter coefficient of this frequency band divided by the power of subband signal backward to subband signal y;
Be formulated promptly:
Step 140 on each frequency band, deducts the product of subband signal and this band-adaptive filter coefficient backward with subband signal forward, again with all difference additions, the voice signal that obtains exporting.
Be formulated promptly:
In sum, by on frequency, dividing band can guarantee in the valley point of each frequency band of output signal all directions of subtend noise source, be distributed in different frequency bands and in the situation of different azimuth in several noise sources, the valley point of different frequency section can be aimed at several noise sources simultaneously, guarantees that noise is inhibited in the output signal.Be example still with background technology institute act situation, should the present invention be arranged after, can make the valley point of low-frequency band aim at window, and projector is aimed in the valley point of the frequency band at fan noise place, therefore can take into account the noise source of both direction simultaneously.
The microphone that the present invention is not limited to use unidirectivity is to obtaining forward and directive property signal backward, adopts the non-directive system and utilizes the operational method of system shown in Figure 1 also can obtain forward and directive property signal backward.But adopt the microphone of unidirectivity right, to microphone require lowly, do not need specially to select a kind of microphone of model right, big reduction can not take place because of the microphone changes of properties in effect yet; In addition, can realize in quite little bulk that range of application enlarges, low to the manufacturing accuracy requirement, can improve yield rate greatly.
System of the present invention also can adopt analog element to realize.
Claims (5)
1, a kind of sub-band adaptive valley point noise-reduction method may further comprise the steps:
(a) the microphone group is picked up directive property signal and directive property signal forward backward respectively, and the signal energy that system receives in different directions is variant, and the direction of institute's received energy minimum is exactly the valley point;
(b) be decomposed into a plurality of subband signals forward and a plurality of subband signal backward with directive property signal backward in identical frequency band division mode forward with described;
(c) to each frequency band, with subband signal forward and backward the cross correlation function of subband signal obtain the adaptive filter coefficient of this frequency band divided by the power of subband signal backward;
(d) on each frequency band, deduct the product of subband signal and this band-adaptive filter coefficient backward with subband signal forward, again with all difference additions, the voice signal that obtains handling.
2, the method for claim 1 is characterized in that, forward described and backward the directive property signal be digital sampled signal.
3, a kind of sub-band adaptive valley point noise reduction system comprises microphone group, sef-adapting filter, totalizer, it is characterized in that, also comprises two analysis filterbank, a synthesis filter group, and described sef-adapting filter and totalizer have a plurality of, wherein:
Described microphone group is used for pick-up of acoustic signals, produces forward and directive property signal backward, and the signal energy that system receives in different directions is variant, and the direction of institute's received energy minimum is exactly the valley point;
Described two analysis filterbank are respectively applied for forward, the directive property signal is decomposed into a plurality of subband signals forward and a plurality of subband signal backward by identical frequency band division mode backward;
Described a plurality of sef-adapting filter, the subband signal backward that is respectively applied for frequency band multiply by the auto adapted filtering coefficient, export backward subband and adjust signal, described auto adapted filtering coefficient equal this frequency band forward subband signal and backward the cross correlation function of subband signal divided by the power of subband signal backward;
Described a plurality of totalizer, the subband signal forward that is respectively applied for each frequency band deducts subband adjustment signal and output backward;
Described synthesis filter group is used for the output of all totalizers being added up the signal that obtains handling.
4, system as claimed in claim 3 is characterized in that, comprises sampling and AD conversion unit in the described microphone group, and the analog acoustic signal that is used for picking up is converted to digital sampled signal.
5, system as claimed in claim 3 is characterized in that, described microphone group is that unidirectional microphines is right.
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| CN101587712B (en) * | 2008-05-21 | 2011-09-14 | 中国科学院声学研究所 | Directional speech enhancement method based on small microphone array |
| US8538749B2 (en) * | 2008-07-18 | 2013-09-17 | Qualcomm Incorporated | Systems, methods, apparatus, and computer program products for enhanced intelligibility |
| CN101430882B (en) * | 2008-12-22 | 2012-11-28 | 无锡中星微电子有限公司 | Method and apparatus for restraining wind noise |
| CN102348151B (en) | 2011-09-10 | 2015-07-29 | 歌尔声学股份有限公司 | Noise canceling system and method, intelligent control method and device, communication equipment |
| CN104200817B (en) * | 2014-07-31 | 2017-07-28 | 广东美的制冷设备有限公司 | Sound control method and system |
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| CA2488918A1 (en) * | 1994-08-10 | 1996-02-22 | Qualcomm Incorporated | Method and apparatus for selecting an encoding rate in a variable rate vocoder |
| CN1194068A (en) * | 1995-07-06 | 1998-09-23 | 相干通讯系统公司 | Spectral noise compensation for echo cancellation |
| US6144937A (en) * | 1997-07-23 | 2000-11-07 | Texas Instruments Incorporated | Noise suppression of speech by signal processing including applying a transform to time domain input sequences of digital signals representing audio information |
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| CA2488918A1 (en) * | 1994-08-10 | 1996-02-22 | Qualcomm Incorporated | Method and apparatus for selecting an encoding rate in a variable rate vocoder |
| CN1194068A (en) * | 1995-07-06 | 1998-09-23 | 相干通讯系统公司 | Spectral noise compensation for echo cancellation |
| US6144937A (en) * | 1997-07-23 | 2000-11-07 | Texas Instruments Incorporated | Noise suppression of speech by signal processing including applying a transform to time domain input sequences of digital signals representing audio information |
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