CN101140760A - Sound signal collecting and processing system and method thereof - Google Patents
Sound signal collecting and processing system and method thereof Download PDFInfo
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Abstract
The invention discloses a system and method for sound signal acquisition and processing, which comprises a sound signal acquisition module, a sound signal pretreatment module, a beam forming module, a D/A conversion module and a noise elimination module and is used for acquiring the frequency of the beam outputted by beam forming module which is the same as that of the first spectrum component, and then send it to the D/A conversion module. The sound pressure difference of the said first spectrum component on the pick-up surface of the sound sensing element is at the predetermined value range. The said sound signal acquisition module includes a plurality of sound sensing elements whose pick-up surfaces vary with the sound source need to collect in distance. By making the sound sensing elements coaxial and co-rotated and setting a certain interval distance, the invention filters out the spectrum components whose sound pressure difference is not in the predetermined value range to eliminate the effect caused by noise inside and outside the beam and improve the quality for signal processing.
Description
Technical Field
The invention relates to processing of sound signals, in particular to a sound signal acquisition device, a sound signal acquisition processing device and a sound signal acquisition processing method capable of eliminating background noise.
Background
At present, a mobile phone, a PDA, a telephone, an earphone with a sound signal acquisition device and the like all need to use the sound signal acquisition device, and the sound signal acquisition device is used for acquiring a voice signal to be sent and then sending the voice signal to the other party.
However, when a user is in a noisy environment, the background noise is high in decibels, the signal collected by the sound signal collecting device includes background noise, and it is difficult to hear the original voice when the collected signal is decoded and output.
From data analysis, it can be known that the user's voice can only be heard from the restored voice if the voice output by the user is above 15dB above background noise. When user's pronunciation is less than background noise more than 15dB, if want clear discernment user's pronunciation in the signal that restores, under the normal conditions, the person of talkbacking can only shout with high voice, and the effect is very limited like this, firstly because the background noise decibel very high reason, the user is very difficult to surpass its 15dB, and simultaneously, sound signal collection system's gain is limited, and user's sound too big will cause the distortion, still can't clear discernment user's voice in the signal that restores out.
Currently, the sound signal collection device includes a dual-microphone solution, which uses a beam forming algorithm to eliminate the noise outside the beam, and the microphones are arranged side by side as shown in fig. 1.
The algorithm for realizing beamforming by the microphone array in a side-by-side manner shown in fig. 1 adopts an adaptive beamforming algorithm based on frequency domain LMS (Least Mean Square), and the processing procedure thereof is as follows, and includes:
step S1, after converting the sound signal collected by the microphone array into a digital speech signal x (n), performing Fast Fourier Transform (FFT) on the input signal x (n), and using a matrix to represent the input signal x (n) as r (n) = Wx (n), where: w is a frequency domain transform matrix, unitary matrix, which is:
and in N-point FFT of the array receiving signals, N is the number of the microphones in the array, and if the number of the microphones is not an integral power of 2, a zero filling method is adopted.
Step S2, processing the input signal after FFT by using LMS algorithm:
y(n)＝V T (n)r(n)
wherein: v is the weight vector in the LMS algorithm.
e(n)＝d(n)-y(n)
Wherein: d is a training sequence.
V(n+1)＝V(n)+2ae(n)r * (n)
Wherein: a is a learning step.
And adding samples, and circularly executing the steps to finally obtain the beam, wherein the finally obtained beam eliminates noise outside the beam.
However, the adaptive beamforming algorithm based on frequency domain LMS (Least Mean Square) described above cannot eliminate noise in the beam.
Disclosure of Invention
The invention aims to provide a sound signal acquisition and processing system and a sound signal acquisition and processing method, which can eliminate noises in a wave beam and noises outside the wave beam.
In order to achieve the above object, the present invention provides an audio signal acquisition processing system, which includes an audio signal acquisition module, an audio signal preprocessing module, a beam forming module, a D/a conversion module, and a noise cancellation module, where the noise cancellation module is configured to acquire a portion of a beam output by the beam forming module, where the frequency is the same as that of a first spectral component, and send the portion of the beam to the D/a conversion module, and a sound pressure difference of the first spectral component on a sound pickup surface of a sound sensing element is within a predetermined value range of the sound pressure difference;
the sound signal acquisition module comprises a plurality of sound sensing elements, and the distances between the pickup surfaces of different sound sensing elements and a sound source to be acquired are different.
In the above sound signal collecting and processing system, the sound sensing elements are coaxially and equidirectionally arranged, and the adjacent sound sensing elements are spaced by a predetermined distance.
The above-mentioned sound signal collecting and processing system, wherein the sound signal preprocessing module specifically includes:
the differential amplifier is correspondingly connected with the sound sensing element;
and the A/D conversion module is used for converting the signals output by the differential amplifier into digital voice signals and outputting the digital voice signals to the beam forming module.
In the sound signal collecting and processing system, the beam forming module is specifically configured to calculate the beam by using an adaptive beam forming algorithm based on frequency domain least mean square.
The above-mentioned sound signal acquisition processing system, wherein, still include:
and the correction module is arranged between the sound signal preprocessing module and the beam forming module and is used for compensating the electrical performance difference between the sound sensing elements.
The above-mentioned sound signal acquisition processing system, wherein, still include:
and the switching module is used for selecting the signal output by one sound sensing element to be sent to the sound processing chip or selecting the output signals of all the sound sensing elements to be sent to the corresponding sound signal preprocessing module.
In order to better achieve the above object, the present invention further provides an acoustic signal collecting and processing method, wherein after the acoustic signals are collected by a plurality of acoustic sensing elements having different distances between the sound collecting surface and the sound source to be collected, after a/D conversion and beam forming, a part of the beam having the same frequency as a first spectral component is obtained, and a part of the beam having the same frequency as the first spectral component is D/a converted and output, and a sound pressure difference of the first spectral component on the sound collecting surface of the acoustic sensing element is within a predetermined sound pressure difference range.
The sound signal collecting and processing method specifically comprises the following steps:
step S1, collecting sound signals by a plurality of sound sensing elements with different distances between a sound pickup surface and a sound source to be collected;
s2, respectively converting the sound signals collected by the plurality of paths into digital voice signals;
s3, obtaining beams from the multi-channel digital voice signals by using a beam forming algorithm;
and S4, acquiring the part with the same frequency as the first frequency spectrum component in the wave beam, and outputting the part after D/A conversion.
In the sound signal collecting and processing method, step S4 specifically includes:
step S41, acquiring an output signal of the sound sensing element;
step S42, calculating the sound pressure difference of the frequency spectrum component of the output signal on the sound sensing element;
step S43, acquiring a frequency spectrum component of which the sound pressure difference on the sound sensing element is in a preset value range of the sound pressure difference;
step S44, taking out the part with the same frequency as the first frequency spectrum component in the wave beam, performing D/A conversion and outputting; or
Step S44', filtering out the part of the wave beam with the frequency different from that of the first frequency spectrum component, and outputting the rest part after D/A conversion;
in the above method for collecting and processing a sound signal, between step S1 and step S2, the method further includes:
and S5, respectively carrying out differential amplification processing on the multiple collected sound signals.
In the sound signal collecting and processing method, the beam forming algorithm is a frequency domain-based least mean square adaptive beam forming algorithm.
In the above sound signal collecting and processing method, before the step S1, the method further includes:
and S6, judging whether the noise elimination processing needs to be started or not, if so, entering the step S1, and otherwise, selecting a signal output by the sound sensing element and sending the signal to the sound processing chip.
According to the sound signal acquisition processing system and method, the sound sensing elements are coaxially arranged in the same direction and are arranged at a certain distance, so that after the wave beam is formed, the frequency spectrum component with the sound pressure difference in the wave beam not in a preset value range is filtered, the influence of noise in the wave beam and outside the wave beam is effectively eliminated, and the quality of signal processing is improved.
Drawings
FIG. 1 is a schematic diagram of an array of microphones arranged in a side-by-side manner;
FIG. 2 is a schematic structural diagram of an audio signal acquisition processing system according to an embodiment of the present invention;
FIG. 3 is a schematic view of another arrangement of the sound sensing element of the sound signal collection module according to the present invention;
fig. 4 is a schematic structural diagram of a sound signal preprocessing module in the sound signal collecting and processing system according to the embodiment of the present invention;
fig. 5 is a schematic structural diagram of a sound signal collecting and processing system according to an embodiment of the present invention;
fig. 6 is a schematic flow chart of a sound signal collecting and processing method according to an embodiment of the present invention.
Detailed Description
The present invention will be described in detail with reference to the accompanying drawings.
As shown in fig. 2, the sound signal collection processing system of the present invention includes:
the sound signal acquisition module 21 is used for acquiring sound signals, and comprises a plurality of sound sensing elements, wherein the distances between the sound pickup surface of the sound signal acquisition module and a sound source to be acquired are different, so that the sound signals emitted by the sound source have different sound pressures on different sound sensing elements;
the sound signal preprocessing module 22 is in number connection with the sound signal collecting module and is used for respectively converting the sound signals collected by the sound sensing elements into digital sound signals; the sound signal preprocessing module 22 respectively converts the sound signals collected in multiple paths and respectively outputs the sound signals to the beam forming module 23;
a beam forming module 23, configured to obtain a beam according to the converted multiple channels of digital voice signals and by using a frequency domain LMS (Least Mean Square) -based adaptive beam forming algorithm, and filter a noise signal outside the beam;
a noise elimination module 24 for acquiring and transmitting a part of the beam having the same frequency as a first spectrum component having a sound pressure difference at the sound sensing element within a predetermined sound pressure difference range;
and the D/a conversion module 25 is configured to perform D/a conversion on the signal output by the noise elimination module 24, and send the signal to a corresponding sound processing chip for processing.
Here, the sound sensing elements in the sound signal collecting module may be coaxially and coaxially arranged in the same direction, and adjacent sound sensing elements are spaced apart by a predetermined distance Δ r, so that sound signals emitted from the sound source have different sound pressures at different sound sensing elements (as will be described in detail later).
The sound sensing elements all comprise a sound pickup surface, and the same-direction arrangement refers to the situation that the sound pickup surfaces of the sound sensing elements face the same direction.
Where co-axial co-orientation of the sound sensing elements is a preferred aspect of the invention, it is permissible that the distance between the axes of the sound sensing surfaces of the sound sensing elements be less than a threshold axial distance if the angle between the sound sensing surfaces of the sound sensing elements is less than a threshold angle, as shown in fig. 3.
In the embodiment of the present invention, the sound sensing elements are arranged coaxially and in the same direction, and adjacent sound sensing elements are spaced apart from each other by a predetermined distance, but other arrangements of the sound sensing elements as shown in fig. 3 are also possible as long as the sound signals emitted from the sound source can exhibit different sound pressures on the sound sensing elements.
As shown in fig. 2, the sound signal collecting module 21 includes a plurality of sound sensing elements coaxially and equidirectionally disposed, and adjacent sound sensing elements are spaced apart by a predetermined distance Δ r.
Generally, the farther from a sound source, the smaller the sound intensity of a point, and if the absorption of sound energy by a medium is not considered, the sound intensity of the point is I = W/4 pi r when the point radiates sound energy uniformly to the periphery in a free sound field 2 Wherein I is the sound intensity at a distance r from the point sound source, and W is the point sound source power.
If the S table does not surround the closed surface area of the sound source, the relationship between the sound power W and the sound intensity I is as follows:
wherein I n Is the component of the sound intensity in the normal direction of the infinitesimal area dS.
In the free and semi-free sound fields, the relationship between the sound pressure level and the sound power level of the point sound source is as follows:
L p ＝L w -20lgr-11 (dB)
L p ＝L w -20lgr-8 (dB)
if the distances between the measuring points 1 and 2 and the sound source are r respectively 1 、r 2 Then r is 1 To r 2 The sound pressure level attenuation amount is: for every doubling of the distance, the sound pressure level is attenuated by about 6dB.
In the present invention, with a coaxially and concentrically arranged microphone module, r2= r1+ Δ r, where Δ r is assumed to be 5mm, then:
the following correspondences can be obtained from the above equation (only the correspondences at the respective orders of magnitude are listed here to draw a conclusion):
r 1 10mm, sound pressure on two acoustic sensing elementsThe difference is 3.521825dB;
r 1 50mm, the acoustic pressure difference across the two acoustic sensing elements is 0.827854dB;
r 1 100mm, the acoustic pressure difference between the two acoustic sensing elements is 0.423786dB;
r 1 300mm, the acoustic pressure difference across the two acoustic sensing elements is 0.143572dB;
r 1 500mm, the acoustic pressure difference over the two acoustic sensing elements is 0.086427dB;
r 1 at 1500mm, the acoustic pressure difference across the two acoustic sensing elements is 0.028985dB.
As can be seen from the above correspondence, when r is 1 When the sound emitting point is larger (more than 500 mm), the sound pressure difference formed by the sound emitting point on the two microphones is quite small (more than 500mm, the formed sound pressure difference is less than 0.1 dB).
However, generally, the distance between the sound signal collecting module and the sound source to be collected is not too large, for example, the distance is not more than 200mm in a mobile phone, and the distance is not more than 300mm in a microphone, so that the sound pressure difference (converted into voltage) can be effectively identified basically.
Therefore, the noise elimination module can completely filter out the spectral components with the sound pressure difference in the wave beam outside the preset value range of the sound pressure difference (or extract only the spectral components with the sound pressure difference in the preset value range of the sound pressure difference), so that the noise in the wave beam is filtered out, and a cleaner signal is obtained.
Wherein, the distance of the sound sensing element in the sound signal acquisition module can be set according to the practical application, such as r 1 When the sound pressure difference is larger, the distance between the sound sensing elements can be set to be slightly larger, otherwise, the distance between the sound sensing elements can be set to be slightly smaller, of course, a sound pressure difference preset value storage module can also be arranged on the noise elimination module 24 and used for modifying and storing the sound pressure difference preset value, and the noise elimination module 24 filters noise according to the stored sound pressure difference preset value. Is collected if necessaryWhen the sound source is far away, the preset value of the sound pressure difference can be set to be small, so that the required sound can be collected (or not filtered), and otherwise, the preset value of the sound pressure difference can be set to be large. Certainly, the sound pressure difference preset value storage module can also be used for storing a plurality of sound pressure difference preset values, and a user selects the corresponding sound pressure difference preset value according to the collected sound source distance to ensure that the required sound is collected and the unnecessary sound (noise) is filtered.
Meanwhile, when the routing of the sound signal collection module is long, common mode noise may be caused, and therefore, as shown in fig. 4, the sound signal preprocessing module of the present invention specifically includes:
the differential amplifier is used for filtering and suppressing common-mode noise;
and the A/D conversion module is used for converting the signal output by the differential amplifier into a digital voice signal.
The beam forming module calculates beams by adopting a frequency domain LMS-based adaptive beam forming algorithm and filters noise signals outside the beams.
Meanwhile, the collected signals of a plurality of sound sensing elements are adopted for processing, so that in order to achieve better effect, the sound signal collecting and processing system also arranges a correction module between the sound signal preprocessing module and the beam forming module for compensating the difference of electrical performance between the sound sensing elements.
Meanwhile, considering that the sound signal collecting and processing system is applied to any occasion and needs to consume energy, but under the condition of low background noise, clear voice reduction can be realized without filtering the background noise, and at the moment, in order to reduce power consumption and prolong service time, a signal of a sound sensing element can be adopted, so the sound signal collecting and processing system further comprises:
the switching module is configured to select a signal output by one sound sensing element and send the signal to a sound processing chip (e.g., a baseband microphone input circuit of a mobile phone) for subsequent processing, or send output signals of all the sound sensing elements to a sound signal preprocessing module for operations such as differential amplification and a/D conversion.
The switching operation of the switching module can be switched by a user through keys, software icons or menu options, and the implementation thereof is well known to those skilled in the art and will not be described herein again.
The noise cancellation module specifically performs the following operations:
firstly, acquiring an output signal of a sound sensing element;
calculating the acoustic pressure difference of the corresponding spectral components over the acoustic sensing element;
acquiring the frequency spectrum component of the sound pressure difference on the sound sensing element within a preset value range of the sound pressure difference;
and correspondingly acquiring the same-frequency signals in the wave beams according to the frequency spectrum components of the sound pressure difference on the sound sensing element in the preset value range of the sound pressure difference.
Finally, a preferred embodiment of the sound signal collection and processing system of the present invention is shown in fig. 5, which illustrates a sound signal collection and processing system using two sound sensing elements.
The following describes the sound signal collecting and processing method of the present invention with reference to the accompanying drawings.
As shown in fig. 6, the sound signal collecting and processing method of the present invention includes the following steps:
and step 65, the D/A conversion module performs D/A conversion on the signal output by the noise elimination module and outputs the signal.
step 641, the noise elimination module obtains an output signal of the sound sensing element;
642, the noise cancellation module calculates an acoustic pressure difference of the corresponding spectral component over the acoustic sensing element;
step 643, the noise elimination module obtains a frequency spectrum component of the sound pressure difference on the sound sensing element in a preset value range of the sound pressure difference;
step 644, the noise elimination module correspondingly acquires the same frequency signal in the beam according to the frequency spectrum component of the sound pressure difference on the sound sensing element within the preset value range of the sound pressure difference and outputs the same frequency signal.
Meanwhile, the method also comprises the following steps between the step 61 and the step 62:
and step 66, carrying out differential amplification processing on the output signal of the sound sensing element by the differential amplifier to eliminate common-mode noise.
Meanwhile, the method also comprises the following steps between the step 62 and the step 63:
step 67, the calibration module compensates for the electrical performance differences between the acoustic sensing elements, and specifically performs the following operations:
and (3) respectively performing equal sound pressure and full frequency band scanning input on each microphone, and respectively acquiring respective frequency spectrum and sampling volume. And comparing the microphone specifications, and compensating the difference positions. Thereby achieving a correction of the dissimilarity between the microphones.
Meanwhile, step 61 of the present invention further comprises:
and 68, judging whether the noise elimination processing needs to be started, if so, entering the step 61, otherwise, selecting a signal output by the sound sensing element and sending the signal to a sound processing chip (such as a baseband microphone input circuit of a mobile phone).
The foregoing is only a preferred embodiment of the present invention, and it should be noted that, for those skilled in the art, various modifications and decorations can be made without departing from the principle of the present invention, and these modifications and decorations should also be regarded as the protection scope of the present invention.
Claims (12)
1. A sound signal acquisition processing system comprises a sound signal acquisition module, a sound signal preprocessing module, a beam forming module and a D/A conversion module, and is characterized by further comprising:
the noise elimination module is used for acquiring a part of the wave beam output by the wave beam forming module, wherein the frequency of the part is the same as that of a first frequency spectrum component, and sending the part to the D/A conversion module, and the sound pressure difference of the first frequency spectrum component on the sound pickup surface of the sound sensing element is in a preset value range of the sound pressure difference;
the sound signal acquisition module comprises a plurality of sound sensing elements, and the distances between sound pickup surfaces of different sound sensing elements and a sound source to be acquired are different.
2. The sound signal collection processing system of claim 1, wherein the sound sensing elements are coaxially and co-directionally arranged, and adjacent sound sensing elements are spaced apart by a predetermined distance.
3. The system for acquiring and processing a sound signal according to claim 1, wherein the sound signal preprocessing module specifically comprises:
the differential amplifier is correspondingly connected with the sound sensing element;
and the A/D conversion module is used for converting the signals output by the differential amplifier into digital voice signals and outputting the digital voice signals to the beam forming module.
4. The sound signal collection processing system of claim 1, wherein the beamforming module employs an adaptive beamforming algorithm based on frequency domain least mean square.
5. The sound signal collection processing system according to claim 1, further comprising:
and the correction module is arranged between the sound signal preprocessing module and the beam forming module and is used for compensating the electrical performance difference between the sound sensing elements.
6. The sound signal collection processing system according to claim 1,2, 3, 4, or 5, further comprising:
and the switching module is used for selecting the signal output by one sound sensing element to be sent to the sound processing chip or selecting the output signals of all the sound sensing elements to be sent to the corresponding sound signal preprocessing module.
7. A sound signal collecting and processing method is characterized in that after sound signals are collected by a plurality of sound sensing elements with different distances between a sound collecting surface and a sound source to be collected, after A/D conversion and beam forming, a part with the same frequency as a first frequency spectrum component in a beam is obtained, and the part with the same frequency as the first frequency spectrum component in the beam is subjected to D/A conversion and then output, wherein the sound pressure difference of the first frequency spectrum component on the sound collecting surface of the sound sensing elements is in a preset sound pressure difference value range.
8. The sound signal collection processing method according to claim 7, specifically comprising:
step S1, collecting sound signals by a plurality of sound sensing elements with different distances between a sound pickup surface and a sound source to be collected;
s2, respectively converting the sound signals collected by the plurality of paths into digital voice signals;
s3, obtaining beams from the multi-channel digital voice signals by using a beam forming algorithm;
and S4, acquiring the part with the same frequency as the first frequency spectrum component in the wave beam, and outputting the part after D/A conversion.
9. The sound signal collection processing method according to claim 8, wherein the step S4 specifically includes:
step S41, acquiring an output signal of the sound sensing element;
step S42, calculating the sound pressure difference of the frequency spectrum component of the output signal on the sound sensing element;
step S43, acquiring a frequency spectrum component of the output signal, wherein the sound pressure difference of the sound sensing element is in a preset value range of the sound pressure difference;
step 844, extracting the part with the same frequency as the first frequency spectrum component in the wave beam, performing D/a conversion, and outputting; or
And step S44', filtering out the part of the wave beam with the frequency different from that of the first frequency spectrum component, and outputting the rest part after D/A conversion.
10. The sound signal collection processing method according to claim 8 or 9, further comprising, between the step S1 and the step S2:
and S5, respectively carrying out differential amplification processing on the multiple collected sound signals.
11. The sound signal collection processing method according to claim 8 or 9, wherein the beamforming algorithm is a frequency domain-based least mean square adaptive beamforming algorithm.
12. The sound signal collection processing method according to claim 8 or 9, wherein the step S1 further comprises:
and S6, judging whether the noise elimination processing needs to be started or not, if so, entering the step S1, and otherwise, selecting a signal output by the sound sensing element and sending the signal to the sound processing chip.
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| CN105049802A (en) * | 2015-07-13 | 2015-11-11 | 深圳警翼数码科技有限公司 | Speech recognition law-enforcement recorder and recognition method thereof |
| CN105049802B (en) * | 2015-07-13 | 2018-06-19 | 深圳警翼智能科技股份有限公司 | A kind of speech recognition law-enforcing recorder and its recognition methods |
| CN106708041A (en) * | 2016-12-12 | 2017-05-24 | 西安Tcl软件开发有限公司 | Intelligent sound box and intelligent sound box directional movement method and device |
| CN108174320A (en) * | 2018-01-29 | 2018-06-15 | 广东思派康电子科技有限公司 | A kind of adaptive duct active noise reduction earphone and adaptive duct active denoising method |
| CN108174320B (en) * | 2018-01-29 | 2019-11-29 | 广东思派康电子科技有限公司 | A kind of adaptive ear canal active noise reduction earphone and adaptive ear canal active denoising method |
| CN109670257A (en) * | 2018-12-26 | 2019-04-23 | 国家电网有限公司 | A kind of converter station sound field emulation mode and system |
| CN109670257B (en) * | 2018-12-26 | 2023-05-05 | 国家电网有限公司 | Sound field simulation method and system for converter station |
| CN111479180A (en) * | 2019-01-24 | 2020-07-31 | Oppo广东移动通信有限公司 | Pickup control method and related product |
| CN111479180B (en) * | 2019-01-24 | 2022-04-29 | Oppo广东移动通信有限公司 | Pickup control method and related product |
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