CN107180642B - Audio signal correction method, device and equipment - Google Patents
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Abstract
The invention provides an audio signal correction method, an audio signal correction device and audio signal correction equipment, wherein the method comprises the following steps: acquiring a plurality of paths of audio signals acquired by a microphone array; the multi-channel audio signals comprise reference audio signals collected by the reference microphone and common audio signals collected by the common microphone; determining a difference parameter between the normal audio signal and the reference audio signal based on a predetermined sample difference parameter; and correcting the common audio signal based on the difference parameter to obtain a corrected audio signal matched with the reference audio signal. The invention can eliminate the difference between the audio signals collected by each microphone to a great extent and improve the performance of the subsequent speech enhancement processing link.
Description
Technical Field
The present invention relates to the field of signal processing technologies, and in particular, to a method, an apparatus, and a device for correcting an audio signal.
Background
The microphone array has better anti-noise performance, anti-interference performance and anti-reverberation performance due to the adoption of the spatial filtering technology, gradually replaces the traditional single microphone, and is widely applied to the technical fields of far-field voice acquisition and the like.
Subsequent speech enhancement processing links (such as sound source positioning, beam forming and the like) of the microphone array require common audio signals of all channels to be matched with each other (such as amplitude and time delay matching) so as to improve the speech enhancement performance to the maximum extent.
However, in practical applications, microphones of each channel often have a large dispersion degree, and characteristics of electronic components of each signal acquisition channel are affected by external conditions such as temperature and humidity, so that a large difference exists between channels, and common audio signals of each channel are not matched, thereby affecting performance of a subsequent speech enhancement processing link.
Disclosure of Invention
In view of this, the present invention provides an audio signal correction method, apparatus and device, so as to match the signals collected by each channel of a microphone array, and improve the performance of the subsequent speech enhancement processing link.
In a first aspect, an embodiment of the present invention provides an audio signal correction method, where the method is used to process an audio signal collected by a microphone array, where the microphone array includes a reference microphone and one or more ordinary microphones; the method comprises the following steps:
acquiring a plurality of paths of audio signals collected by the microphone array; the multi-channel audio signals comprise reference audio signals collected by the reference microphone and common audio signals collected by the common microphone;
determining a difference parameter between the normal audio signal and the reference audio signal based on a predetermined sample difference parameter;
and correcting the common audio signal based on the difference parameter to obtain a corrected audio signal matched with the reference audio signal.
Optionally, the method for determining the sample difference parameter includes:
acquiring multi-path sample audio signals of a sample sound source collected by the microphone array; wherein the multi-path sample audio signal comprises a reference sample audio signal collected by the reference microphone and a common sample audio signal collected by the common microphone;
and respectively determining the sample difference parameters of each path of sample audio signal in the common sample audio signal and the reference sample audio signal.
Optionally, the separately determining the sample difference parameter between each of the normal sample audio signals and the reference sample audio signal includes:
calculating a mean square value of the reference sample audio signal;
calculating a mean square value of the common sample audio signal;
calculating the cross-correlation function of the reference sample audio signal and each path of common sample audio signal;
carrying out interpolation operation on the cross-correlation function to obtain an interpolation cross-correlation function;
and respectively determining the sample difference parameters of the common sample audio signals and the reference sample audio signal according to the mean square value of the reference sample audio signal, the mean square value of the common sample audio signal and the interpolation cross-correlation function.
Optionally, the sample difference parameter comprises a delay value;
the determining the sample difference parameters of the common sample audio signals and the reference sample audio signal according to the mean square value of the reference sample audio signal, the mean square value of the common sample audio signal and the interpolation cross-correlation function respectively includes:
respectively determining the apparent time delay values of the common sample audio signals and the reference sample audio signals according to the interpolation cross-correlation function;
respectively determining the transmission path time delay difference of each path of common sample audio signal and the reference sample audio signal;
and respectively calculating the time delay values of the common sample audio signals and the reference sample audio signals according to the apparent time delay value and the transmission path time delay difference.
Optionally, the determining the transmission path delay difference between the common sample audio signals and the reference sample audio signal respectively includes:
establishing a three-dimensional rectangular coordinate system by taking the reference microphone as a coordinate origin;
determining a sound source direction vector according to the coordinates of the sample sound source and the distance between the sample sound source and the origin of coordinates; the coordinates of the sample sound source are the corresponding coordinates of the sample sound source in the three-dimensional rectangular coordinate system;
respectively determining the transmission path time delay difference of each path of common sample audio signal and the reference sample audio signal according to the respective coordinate of the common microphone and the sound source direction vector; and the respective coordinates of the common microphones are the respective corresponding coordinates of the common microphones in the three-dimensional rectangular coordinate system.
Optionally, the determining a difference parameter between the normal audio signal and the reference audio signal based on a predetermined sample difference parameter includes:
taking the time delay value of each path of common audio signal and the reference sample audio signal as the time delay value of each path of common audio signal and the reference audio signal;
the correcting the common audio signal based on the difference parameter includes:
and carrying out filtering delay operation on each path of common audio signal according to the delay value of each path of common audio signal and the reference audio signal.
Optionally, the sample difference parameter comprises an amplitude ratio;
the determining the sample difference parameters of the common sample audio signals and the reference sample audio signal according to the mean square value of the reference sample audio signal, the mean square value of the common sample audio signal and the interpolation cross-correlation function respectively includes:
and determining the amplitude ratio of each path of common sample audio signal to the reference sample audio signal according to the ratio of the mean square value of each path of common sample audio signal to the mean square value of the reference sample audio signal.
Optionally, the determining a difference parameter between the normal audio signal and the reference audio signal based on a predetermined sample difference parameter includes:
taking the amplitude ratio of each path of common audio signal to the reference sample audio signal as the amplitude ratio of each path of common audio signal to the reference audio signal;
the correcting the common audio signal based on the difference parameter includes:
and performing division operation according to the amplitude ratio of each path of common audio signal to the reference audio signal and the common audio signals.
In a second aspect, an embodiment of the present invention provides an audio signal correction apparatus, where the apparatus is configured to process an audio signal collected by a microphone array, where the microphone array includes a reference microphone and one or more ordinary microphones; the device comprises:
the signal acquisition module is used for acquiring a plurality of paths of audio signals acquired by the microphone array; the multi-channel audio signals comprise reference audio signals collected by the reference microphone and common audio signals collected by the common microphone;
a parameter determination module for determining a difference parameter between the common audio signal and the reference audio signal based on a predetermined sample difference parameter;
and the signal correction module is used for correcting the common audio signal based on the difference parameter to obtain a corrected audio signal matched with the reference audio signal.
In a third aspect, an embodiment of the present invention provides an electronic device, where the electronic device is configured to process an audio signal collected by a microphone array, where the microphone array includes a reference microphone and one or more ordinary microphones; the electronic device includes:
a processor;
a memory configured to store processor-executable instructions;
wherein the processor is configured to:
acquiring a plurality of paths of audio signals collected by the microphone array; the multi-channel audio signals comprise reference audio signals collected by the reference microphone and common audio signals collected by the common microphone;
determining a difference parameter between the normal audio signal and the reference audio signal based on a predetermined sample difference parameter;
and correcting the common audio signal based on the difference parameter to obtain a corrected audio signal matched with the reference audio signal.
In a fourth aspect, embodiments of the present invention provide a computer-readable storage medium for processing audio signals collected by a microphone array, the microphone array including a reference microphone and one or more ordinary microphones; the storage medium having stored thereon a computer program that, when processed by a processor, implements:
acquiring a plurality of paths of audio signals collected by the microphone array; the multi-channel audio signals comprise reference audio signals collected by the reference microphone and common audio signals collected by the common microphone;
determining a difference parameter between the normal audio signal and the reference audio signal based on a predetermined sample difference parameter;
and correcting the common audio signal based on the difference parameter to obtain a corrected audio signal matched with the reference audio signal.
According to the technical scheme, the audio signal correction method, the device and the equipment provided by the invention have the advantages that the multi-channel audio signals collected by the microphone array are obtained, the difference parameters between the common audio signals and the reference audio signals are determined based on the predetermined sample difference parameters, the common audio signals are corrected based on the difference parameters, the corrected audio signals matched with the reference audio signals are obtained, the difference between the audio signals collected by all the microphones can be eliminated to a great extent, and the performance of a subsequent speech enhancement processing link is improved.
Drawings
FIG. 1 is a flow chart of an embodiment of a method for correcting an audio signal according to the present invention;
FIG. 2 is a flow diagram of an embodiment of the present invention for determining a sample difference parameter;
FIG. 3 is a flow chart of an embodiment of the present invention for determining sample difference parameters for each of a plurality of normal sample audio signals and a reference sample audio signal;
FIG. 4A is a flow chart of another embodiment of a method for correcting an audio signal according to the present invention;
FIG. 4B is a schematic diagram of a three-dimensional microphone array arrangement and sound source location embodiment of the present invention;
FIG. 5 is a block diagram of an embodiment of an audio signal correction apparatus according to the present invention;
FIG. 6 is a block diagram of another embodiment of an audio signal correction apparatus according to the present invention;
fig. 7 is a block diagram of an electronic device according to an embodiment of the present invention.
Detailed Description
In order to make the aforementioned objects, features and advantages of the present invention comprehensible, embodiments accompanied with figures are described in further detail below.
The terminology used herein is for the purpose of describing particular embodiments only and is not intended to be limiting of the application. As used in this application and the appended claims, the singular forms "a", "an", and "the" are intended to include the plural forms as well, unless the context clearly indicates otherwise. It should also be understood that the term "and/or" as used herein refers to and encompasses any and all possible combinations of one or more of the associated listed items.
It is to be understood that although the terms first, second, third, etc. may be used herein to describe various information, such information should not be limited to these terms. These terms are only used to distinguish one type of information from another. For example, first information may also be referred to as second information, and similarly, second information may also be referred to as first information, without departing from the scope of the present application. The word "if" as used herein may be interpreted as "at … …" or "when … …" or "in response to a determination", depending on the context.
FIG. 1 is a flowchart illustrating an embodiment of a method for correcting an audio signal according to the present invention. The method is used for processing audio signals collected by a microphone array, wherein the microphone array comprises a reference microphone and one or more common microphones. As shown in FIG. 1, the method includes the following steps S11-S13:
s11: acquiring a plurality of paths of audio signals collected by the microphone array;
the multi-channel audio signals comprise reference audio signals collected by the reference microphone and common audio signals collected by the common microphone;
in an alternative embodiment, the microphone array may be composed of more than two microphones, and arranged in a three-dimensional space in any three-dimensional array. Wherein, the two-dimensional plane array and the one-dimensional linear array are regarded as the special form of the three-dimensional array.
In an alternative embodiment, the audio signal collected by the microphone array is a digital audio signal after analog-to-digital conversion.
In an alternative embodiment, the reference microphone may be any one of the microphones in the microphone array, for example, the microphone closest to the center of the microphone array may be used as the reference microphone.
In an alternative embodiment, the microphone array may collect the audio signal of the target environment or the target sound source according to actual needs, which is not limited in the embodiment of the present invention.
S12: determining a difference parameter between the normal audio signal and the reference audio signal based on a predetermined sample difference parameter;
in an alternative embodiment, the sample difference parameter is a parameter determined according to a plurality of sample audio signals collected before the microphone array.
In an alternative embodiment, the sample difference parameter may be determined according to the method of the embodiment shown in fig. 2, which will not be described in detail herein.
S13: and correcting the common audio signal based on the difference parameter to obtain a corrected audio signal matched with the reference audio signal.
In an alternative embodiment, the normal audio signal may be corrected according to the determined difference parameter between the normal audio signal and the reference audio signal, so that the corrected normal audio signal matches the reference audio signal.
According to the technical scheme, the audio signal correction method provided by the invention has the advantages that the multi-channel audio signals collected by the microphone array are obtained, the difference parameters between the common audio signals and the reference audio signals are determined based on the predetermined sample difference parameters, the common audio signals are corrected based on the difference parameters, the corrected audio signals matched with the reference audio signals are obtained, the difference between the audio signals collected by each microphone can be eliminated to a great extent, and the performance of the subsequent speech enhancement processing link is improved.
FIG. 2 is a flow diagram of an embodiment of the present invention for determining a sample difference parameter; the present embodiment is exemplified by how to determine the sample difference parameter on the basis of the above embodiments. As shown in fig. 2, the method for determining the sample difference parameter may include the following steps S21-S22:
s21: acquiring multi-path sample audio signals of a sample sound source collected by the microphone array;
wherein the multi-path sample audio signal comprises a reference sample audio signal collected by the reference microphone and a common sample audio signal collected by the common microphone;
in an alternative embodiment, the sample sound source may be located at a fixed position in space. The sample sound source is at least one sound source located at the same position, and the number of the sound sources is not limited.
In an alternative embodiment, when the audio signal of the sample sound source is collected by the microphone array, the sample sound source can be aligned with the center of the microphone array to ensure the quality of audio signal collection.
In an alternative embodiment, the audio signal of the sample sound source may be acquired by the microphone array for a preset time period, and the preset time may be freely set, for example, may be set to be greater than 1 second.
S22: and respectively determining the sample difference parameters of each path of sample audio signal in the common sample audio signal and the reference sample audio signal.
According to the technical scheme, the multi-path sample audio signals of the sample sound source collected by the microphone array are obtained, and the sample difference parameters of each path of sample audio signal in the common sample audio signal and the reference sample audio signal are respectively determined, so that the sample difference parameters can be accurately determined, the difference parameters between the common audio signal and the reference audio signal can be further determined, a basis is provided for subsequently eliminating the difference between the audio signals collected by each path of microphone, and the performance of a subsequent speech enhancement processing link is improved.
FIG. 3 is a flow chart of an embodiment of the present invention for determining sample difference parameters for each of a plurality of normal sample audio signals and a reference sample audio signal; based on the above embodiments, the present embodiment takes an example of how to determine the sample difference parameter between each path of the normal sample audio signal and the reference sample audio signal. As shown in fig. 3, the step S22 of determining the sample difference parameters of the normal sample audio signal and the reference sample audio signal respectively includes steps S31-S34:
s31: calculating the mean square value of the reference sample audio signal, and calculating the mean square value of each path of common sample audio signal; in an alternative embodiment, the M sample audio signals collected by the microphone array are sk(n), k is 1,2, … M, where s is the reference sample audio signal acquired by the reference microphoner(n), then the reference sample audio signal s is calculatedr(n) mean square value Rrr(0)＝E{sr(n)sr(n) }, calculating the k-th common sample audio signal sk(n) mean square value Rkk(0)＝E{sk(n)sk(n)}。
S32: calculating the cross-correlation function of the reference sample audio signal and each path of common sample audio signal;
in an alternative embodiment, the reference sample audio signal s is calculated according to the following formular(n) and sample audio signal s collected by k path microphonek(n) cross-correlation function:
Rrk(τ)＝E{sr(n)sk(n+τ)},τ＝…,0,1,2,3,…； (1.1)
s33: performing interpolation operation according to the cross-correlation function to obtain an interpolation cross-correlation function;
in an alternative embodiment, if the maximum point of the cross-correlation function is: i ═ argmax { R }rk(τ) }, then [ i-0.5, i +0.5 }]Interval pair cross correlation function Rrk(τ) interpolating to obtain an interpolated cross-correlation function:
Rrk(η)＝Rrk(i+t)＝Rrk(τ)·sinc(i+t-τ), (1.2)
wherein t is more than or equal to-0.5 and less than or equal to 0.5, and the product in the formula represents the inner product operation;
s34: and respectively determining sample difference parameters of the common sample audio signals and the reference sample audio signal according to the mean square value of the reference sample audio signal, the mean square value of the common sample audio signals and the interpolation cross-correlation function.
In an alternative embodiment, the sample difference parameter may include a delay value;
accordingly, step S34 may include steps S41-S42:
s41: respectively determining the apparent time delay values of the common sample audio signals and the reference sample audio signals according to the interpolation cross-correlation function;
in an alternative embodiment, the interpolated cross-correlation function R may be cross-correlatedrkThe time delay value corresponding to the maximum point of (η) is taken as sk(n) relative to sr(n) apparent delay value D (k) argmax (R)rk(η)}。
It should be noted that the apparent delay value may refer to a delay value directly calculated from the acquired audio signal, without considering the path delay.
S42: and respectively determining the transmission path time delay difference of each path of common sample audio signal and the reference sample audio signal.
In an alternative embodiment, the step S42 may include the following steps S421 to S424:
s421: establishing a three-dimensional rectangular coordinate system by taking the reference microphone as a coordinate origin;
s422: determining a sound source direction vector according to the coordinates of the sample sound source and the distance between the sample sound source and the origin of coordinates; the coordinates of the sample sound source are the corresponding coordinates of the sample sound source in the three-dimensional rectangular coordinate system;
s423: respectively determining the transmission path time delay difference of each path of common sample audio signal and the reference sample audio signal according to the respective coordinate of the common microphone and the sound source direction vector; and the respective coordinates of the common microphones are the respective corresponding coordinates of the common microphones in the three-dimensional rectangular coordinate system.
In an alternative embodiment, let the kth microphone coordinate position be (x)k,yk,zk) And the sound source position coordinates are (X, Y, Z), calculating the transmission path time delay difference of the kth microphone relative to the reference microphone according to the following formula (1.3):
L(k)＝-(xk×X+yk×Y+zk×Z)/(X×X+Y×Y+Z×Z)0.5； (1.3)
s424: and respectively calculating the time delay values of the common sample audio signals and the reference sample audio signals according to the apparent time delay value and the transmission path time delay difference.
In an alternative embodiment, the apparent delay value D (k) is subtracted by the difference L (k) to obtain sk(n) relative to srDelay value of (n):
d(k)＝D(k)-L(k)； (1.4)
on the basis of the above embodiment, the determining the difference parameter between the normal audio signal and the reference audio signal based on the predetermined sample difference parameter in step S12 may include:
taking the time delay value of each path of common audio signal and the reference sample audio signal as the time delay value of each path of common audio signal and the reference audio signal;
accordingly, the correcting the normal audio signal based on the difference parameter in step S13 may include:
and carrying out filtering delay operation on each path of common audio signal according to the delay value of each path of common audio signal and the reference audio signal.
In an alternative embodiment, the k-th (k ═ 1,2, … M) audio signal s is usedk1(n) convolving with sinc (n + d (k)) to make sk1And (n) delaying-d (k) to compensate the time delay of the k path signal.
In an alternative embodiment, the sample difference parameter may include an amplitude ratio;
accordingly, step S34 may further include:
and determining the amplitude ratio of each path of common sample audio signal to the reference sample audio signal according to the ratio of the mean square value of each path of common sample audio signal to the mean square value of the reference sample audio signal.
In an alternative embodiment, s isk(n) mean square divided by srThe mean square value of (n) is then root-coded to obtain sk(n) relative to sr(n) amplitude ratio:
a(k)＝(Rkk(0)/Rrr(0))0.5； (1.5)
on this basis, the determining the difference parameter between the normal audio signal and the reference audio signal based on the predetermined sample difference parameter in step S12 may include:
and taking the amplitude ratio of each path of common audio signal to the reference sample audio signal as the amplitude ratio of each path of common audio signal to the reference audio signal. Accordingly, the correcting the normal audio signal based on the difference parameter in step S13 may include:
and performing division operation according to the amplitude ratio of each path of common audio signal to the reference audio signal and the common audio signals.
In an alternative embodiment, the delayed k-th (k ═ 1,2, … M) audio signal is divided by a (k) to compensate for the amplitude of the signal.
In an alternative embodiment, the M paths of corrected signals after time delay and amplitude compensation can be output to an external system.
According to the technical scheme, the audio signal correction method provided by the invention determines the difference parameter between the common audio signal and the reference audio signal based on the predetermined sample difference parameter, corrects the common audio signal based on the difference parameter to obtain the corrected audio signal matched with the reference audio signal, can eliminate the difference between the audio signals collected by each microphone to a great extent, and improves the performance of the subsequent speech enhancement processing link.
While, for purposes of simplicity of explanation, the foregoing method embodiments have been described as a series of acts or combination of acts, it will be appreciated by those skilled in the art that the present invention is not limited by the illustrated ordering of acts, as some steps may occur in the same order or concurrently with other steps in accordance with the invention.
Further, those skilled in the art should also appreciate that the embodiments described in the specification are exemplary embodiments and that the acts and modules illustrated are not necessarily required to practice the invention.
The present invention is illustrated below by way of a specific example, which is not intended to limit the scope of the invention.
FIG. 4A is a flow chart of another embodiment of a method for correcting an audio signal according to the present invention; as shown in fig. 4A, the method includes:
a1: sample audio signals of sample sound sources are acquired by a microphone array.
In an alternative embodiment, step A1 may include the following steps A11-A12:
a11: and acquiring audio signals of the sample sound source through the microphone array within a preset time period.
Wherein the sample sound source may be disposed at a fixed position in space.
In an alternative embodiment, the sample sound source refers to at least one sound source located at the same position, and the number of sound sources is not limited.
In an alternative embodiment, the microphone array may be composed of more than two microphones, and arranged in a three-dimensional space in any three-dimensional array. Wherein, the two-dimensional plane array and the one-dimensional linear array are regarded as the special form of the three-dimensional array.
In an alternative embodiment, when the audio signal is collected by the microphone array, the sample sound source can be aligned with the center of the microphone array to ensure the quality of audio signal collection.
In an alternative embodiment, the preset time may be freely set, for example, may be set to be greater than 1 second.
A12: and selecting one microphone in the microphone array as a reference microphone.
In an alternative embodiment, the reference microphone may be any one of the microphones in the microphone array, for example, the microphone closest to the center of the microphone array may be used as the reference microphone, and the signal collected by the reference microphone may be used as the reference sample audio signal.
For example, fig. 4B is a schematic diagram of an embodiment of a three-dimensional microphone array arrangement and sound source location according to the present invention. As shown in fig. 4B, the microphone array includes M microphones, a microphone located at the center of the microphone array is selected as a reference microphone, and a sample sound source is aligned to the center of the microphone array, where the position coordinate of the sound source is (X, Y, Z). On the basis, the sample audio signal of the sample sound source collected by the M microphone arrays is set as sk(n), k is 1,2, … M, where s is the reference sample audio signal of the sample sound source acquired by the reference microphoner(n) setting the kth microphone coordinate position as (x)k,yk,zk)。
A2: and calculating two correction parameters of amplitude ratio and time delay value between the sample audio signals collected by the multi-path microphones and the reference sample audio signals.
In an alternative embodiment, step A2 may include the following steps A21-A29:
a21: computing a reference sample audio signal sr(n) mean square value Rrr(0)＝E{sr(n)sr(n)}；
A22: computing reference sample tonesFrequency signal sr(n) and sample audio signal s collected by k path microphonek(n) cross-correlation function:
Rrk(τ)＝E{sr(n)sk(n+τ)},τ＝…,0,1,2,3,…； (2.1)
a23: remember i ═ argmax { Rrk(τ) }, at [ i-0.5, i +0.5]Interval pair cross correlation function Rrk(τ) interpolating to obtain an interpolated cross-correlation function:
Rrk(η)＝Rrk(i+t)＝Rrk(τ)·sinc(i+t-τ)， (2.2)
wherein t is more than or equal to-0.5 and less than or equal to 0.5, and the product in the formula represents the inner product operation;
a24: r is to berkThe time delay value corresponding to the maximum point of (η) is taken as sk(n) relative to sr(n) apparent delay value D (k) argmax (R)rk(η)}；
Where the apparent delay value refers to a delay value calculated directly from the acquired audio signal without taking into account the path delay.
A25: calculating the transmission path time delay difference of the kth microphone relative to the reference microphone:
L(k)＝-(xk×X+yk×Y+zk×Z)/(X×X+Y×Y+Z×Z)0.5； (2.3)
a26: subtracting L (k) from D (k) to obtain sk(n) relative to srDelay value of (n):
d(k)＝D(k)-L(k)； (2.4)
a27: calculating the k-th common audio signal skMean square value of (n):
Rkk(0)＝E{sk(n)sk(n)}； (2.5)
a28: will sk(n) mean square divided by srThe mean square value of (n) is then root-coded to obtain sk(n) relative to sr(n) amplitude ratio:
a(k)＝(Rkk(0)/Rrr(0))0.5； (2.6)
a29: and outputting M paths of amplitude ratios a (k), k being 1,2,3, … and M and time delay values d (k), k being 1,2,3, … and M.
A3: storing two correction parameters of M paths of amplitude ratio a (k) and time delay value d (k).
In an alternative embodiment, the M paths of amplitude ratio values a (k) and delay values d (k) output in step S29 are stored in a non-volatile memory.
In an alternative embodiment, the two correction parameters, i.e., the amplitude ratio a (k) and the delay value d (k), can be read out from the memory according to a subsequent read command.
A4: and according to the two correction parameters of the M paths of read amplitude ratio values a (k) and the time delay values d (k), carrying out amplitude and time delay correction on the multi-path audio signals of the target sound source acquired by the microphone array.
In an alternative embodiment, step a4 may include:
a41: and acquiring multi-channel audio signals of a target sound source through the microphone array.
In an alternative embodiment, the audio signal of the target sound source collected by the M-path microphone array is set as sk1(n), where k is 1,2, … M, and the reference audio signal s is the audio signal of the target sound source collected by the reference microphoner1(n)。
A42: the k-th (k is 1,2, … M) audio signal sk1(n) convolving with sinc (n + d (k)) to make sk1(n) delay-d (k) to compensate the time delay of the k path signal;
a43: dividing the delayed k-th (k is 1,2, … M) audio signal by a (k) to compensate the amplitude of the signal;
a5: and outputting the corrected signal subjected to the time delay and amplitude compensation to an external system.
In the audio signal correction method of the embodiment, a microphone array is used for collecting multiple paths of sample audio signals of a sample sound source, two correction parameters, namely an amplitude ratio and a time delay value, between a reference sample audio signal and each path of audio signal are calculated, and then after the multiple paths of audio signals of a target sound source are collected through the microphone array, amplitude and time delay correction are respectively carried out on the collected multiple paths of audio signals on the basis of the two correction parameters, namely the calculated amplitude ratio and the calculated time delay value, so that multiple paths of correction signals subjected to time delay and amplitude compensation are obtained, differences among the audio signals collected by each path of microphone can be eliminated to a great extent, and the performance of a subsequent speech enhancement processing link is improved.
FIG. 5 is a block diagram of an embodiment of an audio signal correction apparatus according to the present invention; the device is used for processing audio signals collected by a microphone array, wherein the microphone array comprises a reference microphone and one or more common microphones; as shown in fig. 5, the apparatus includes a signal acquisition module 410, a parameter determination module 420, and a signal correction module 430, wherein:
a signal obtaining module 410, configured to obtain multiple audio signals collected by the microphone array; the multi-channel audio signals comprise reference audio signals collected by the reference microphone and common audio signals collected by the common microphone;
a parameter determination module 420 for determining a difference parameter between the normal audio signal and the reference audio signal based on a predetermined sample difference parameter;
a signal correction module 430, configured to correct the common audio signal based on the difference parameter, so as to obtain a corrected audio signal matched with the reference audio signal.
According to the technical scheme, the audio signal correction device provided by the invention obtains the multiple paths of audio signals collected by the microphone array, determines the difference parameter between the common audio signal and the reference audio signal based on the predetermined sample difference parameter, and corrects the common audio signal based on the difference parameter to obtain the corrected audio signal matched with the reference audio signal, so that the difference between the audio signals collected by each path of microphone can be eliminated to a great extent, and the performance of a subsequent speech enhancement processing link is improved.
FIG. 6 is a block diagram of another embodiment of an audio signal correction apparatus according to the present invention; the signal obtaining module 510, the parameter determining module 520, and the signal correcting module 530 have the same functions as the signal obtaining module 410, the parameter determining module 420, and the signal correcting module 430 in the embodiment shown in fig. 5, and are not described herein again. As shown in fig. 6, on the basis of the above embodiment, the apparatus may further include:
a sample signal acquiring module 540, configured to acquire multiple paths of sample audio signals of a sample sound source acquired by the microphone array; wherein the multi-path sample audio signal comprises a reference sample audio signal collected by the reference microphone and a common sample audio signal collected by the common microphone;
a sample difference determining module 550, configured to determine sample difference parameters of each path of the sample audio signal in the common sample audio signal and the reference sample audio signal, respectively.
In an alternative embodiment, the sample difference determination module 550 may include:
a mean square value calculation unit 551 for calculating a mean square value of the reference sample audio signal;
a correlation function determining unit 552, configured to determine, according to the mean square value of the reference sample audio signal, a cross-correlation function between the reference sample audio signal and each of the common sample audio signals;
an interpolation function determining unit 553, configured to perform interpolation operation according to the cross-correlation function to obtain an interpolation cross-correlation function;
a difference parameter determining unit 554, configured to determine sample difference parameters of the common sample audio signals and the reference sample audio signal according to the mean square value of the reference sample audio signal, the mean square value of the common sample audio signals, and the interpolation cross-correlation function. It should be noted that, for the device embodiment, since it basically corresponds to the method embodiment, relevant portions may be referred to only for the description of the method embodiment, and are not described herein again.
The embodiment of the signal processing device of the invention can be applied to network equipment. The device embodiments may be implemented by software, or by hardware, or by a combination of hardware and software. A software implementation is taken as an example, and a logical means is formed by a processor of the device in which it is located running corresponding computer program instructions in a memory. From a hardware aspect, as shown in fig. 7, it is a hardware structure diagram of a device in which the signal processing apparatus of the present invention is located, and besides the processor, the network interface, the memory, and the nonvolatile memory shown in fig. 7, the device in which the apparatus is located in the embodiment may also generally include common hardware, such as a forwarding chip responsible for processing a packet, and the like; the device may also be a distributed device in terms of hardware structure, and may include multiple interface cards to facilitate expansion of message processing at the hardware level.
The embodiment of the invention also provides a computer-readable storage medium, which is used for processing the audio signals collected by a microphone array, wherein the microphone array comprises a reference microphone and one or more common microphones; the storage medium having stored thereon a computer program, the program when processed by a processor implementing:
acquiring a plurality of paths of audio signals collected by the microphone array; the multi-channel audio signals comprise reference audio signals collected by the reference microphone and common audio signals collected by the common microphone;
determining a difference parameter between the normal audio signal and the reference audio signal based on a predetermined sample difference parameter;
and correcting the common audio signal based on the difference parameter to obtain a corrected audio signal matched with the reference audio signal.
The embodiments in the present specification are all described in a progressive manner, each embodiment focuses on differences from the common embodiment, and the same and similar parts among the embodiments may be referred to each other. For the device embodiment, since it is basically similar to the method embodiment, the description is simple, and for the relevant points, refer to the partial description of the method embodiment.
The above description is only for the purpose of illustrating the preferred embodiments of the present invention and is not to be construed as limiting the invention, and any modifications, equivalents, improvements and the like made within the spirit and principle of the present invention should be included in the scope of the present invention.
Claims (9)
1. An audio signal correction method is characterized in that the method is used for processing audio signals collected by a microphone array, and the microphone array comprises a reference microphone and one or more ordinary microphones; the method comprises the following steps:
acquiring a plurality of paths of audio signals collected by the microphone array; the multi-channel audio signals comprise reference audio signals collected by the reference microphone and common audio signals collected by the common microphone;
determining a difference parameter between the normal audio signal and the reference audio signal based on a predetermined sample difference parameter;
correcting the common audio signal based on the difference parameter to obtain a corrected audio signal matched with the reference audio signal;
the method further comprises the following steps:
acquiring multi-path sample audio signals of a sample sound source collected by the microphone array; wherein the multi-path sample audio signal comprises a reference sample audio signal collected by the reference microphone and a common sample audio signal collected by the common microphone;
respectively determining sample difference parameters of each path of sample audio signal in the common sample audio signal and the reference sample audio signal;
the determining the sample difference parameters of each path of sample audio signal in the common sample audio signal and the reference sample audio signal respectively includes:
calculating a mean square value of the reference sample audio signal;
calculating a mean square value of the common sample audio signal;
calculating a cross-correlation function of the reference sample audio signal and each path of sample audio signal;
carrying out interpolation operation on the cross-correlation function to obtain an interpolation cross-correlation function;
and respectively determining the sample difference parameters of the sample audio signals and the reference sample audio signal according to the mean square value of the reference sample audio signal, the mean square value of the common sample audio signal and the interpolation cross-correlation function.
2. The method of claim 1, wherein the sample difference parameter comprises a delay value;
the determining the sample difference parameters of the sample audio signals and the reference sample audio signal according to the mean square value of the reference sample audio signal, the mean square value of the common sample audio signal and the interpolation cross-correlation function respectively includes:
respectively determining apparent time delay values of the sample audio signals and the reference sample audio signal according to the interpolation cross-correlation function;
respectively determining the transmission path time delay difference of each path of sample audio signal and the reference sample audio signal;
and respectively calculating the time delay values of the audio signals of each path of sample and the audio signals of the reference sample according to the apparent time delay value and the time delay difference of the transmission path.
3. The method according to claim 2, wherein the separately determining the transmission path delay difference between the sample audio signals and the reference sample audio signal comprises:
establishing a three-dimensional rectangular coordinate system by taking the reference microphone as a coordinate origin;
determining a sound source direction vector according to the coordinates of the sample sound source and the distance between the sample sound source and the origin of coordinates; the coordinates of the sample sound source are the corresponding coordinates of the sample sound source in the three-dimensional rectangular coordinate system;
respectively determining the transmission path time delay difference of each path of sample audio signal and the reference sample audio signal according to the respective coordinate of the common microphone and the sound source direction vector; and the respective coordinates of the common microphones are the respective corresponding coordinates of the common microphones in the three-dimensional rectangular coordinate system.
4. The method of claim 2, wherein determining the difference parameter between the normal audio signal and the reference audio signal based on a predetermined sample difference parameter comprises:
taking the time delay value of each path of sample audio signal and the reference sample audio signal as the time delay value of each path of common audio signal and the reference audio signal;
the correcting the common audio signal based on the difference parameter includes:
and carrying out filtering delay operation on each path of common audio signal according to the delay value of each path of common audio signal and the reference audio signal.
5. The method of claim 1, wherein the sample difference parameter comprises an amplitude ratio value;
the determining the sample difference parameters of the sample audio signals and the reference sample audio signal according to the mean square value of the reference sample audio signal, the mean square value of the common sample audio signal and the interpolation cross-correlation function respectively includes:
and determining the amplitude ratio of each path of sample audio signal to the reference sample audio signal according to the ratio of the mean square value of each path of sample audio signal to the mean square value of the reference sample audio signal.
6. The method of claim 5, wherein determining the difference parameter between the normal audio signal and the reference audio signal based on a predetermined sample difference parameter comprises:
taking the amplitude ratio of each path of sample audio signal to the reference sample audio signal as the amplitude ratio of each path of common audio signal to the reference audio signal;
the correcting the common audio signal based on the difference parameter includes:
and performing division operation according to the amplitude ratio of each path of common audio signal to the reference audio signal and the common audio signals.
7. An audio signal correction device, characterized in that the device is used for processing audio signals collected by a microphone array, wherein the microphone array comprises a reference microphone and one or more ordinary microphones; the device comprises:
the signal acquisition module is used for acquiring a plurality of paths of audio signals acquired by the microphone array; the multi-channel audio signals comprise reference audio signals collected by the reference microphone and common audio signals collected by the common microphone;
a parameter determination module for determining a difference parameter between the common audio signal and the reference audio signal based on a predetermined sample difference parameter;
the signal correction module is used for correcting the common audio signal based on the difference parameter to obtain a corrected audio signal matched with the reference audio signal;
the device further comprises:
the sample signal acquisition module is used for acquiring a plurality of paths of sample audio signals of a sample sound source acquired by the microphone array; wherein the multi-path sample audio signal comprises a reference sample audio signal collected by the reference microphone and a common sample audio signal collected by the common microphone;
a sample difference determining module, configured to determine sample difference parameters of each path of sample audio signal in the common sample audio signal and the reference sample audio signal respectively;
the sample difference determination module further comprises:
a mean square value calculation unit for calculating a mean square value of the reference sample audio signal;
a correlation function determining unit, configured to determine a cross-correlation function between the reference sample audio signal and each of the channels of sample audio signals according to a mean square value of the reference sample audio signal;
an interpolation function determining unit, configured to perform interpolation operation according to the cross-correlation function to obtain an interpolation cross-correlation function;
and the difference parameter determining unit is used for respectively determining the sample difference parameters of the sample audio signals of each path and the reference sample audio signal according to the mean square value of the reference sample audio signal, the mean square value of the sample audio signals of each path and the interpolation cross-correlation function.
8. An electronic device for processing audio signals collected by a microphone array, the microphone array comprising a reference microphone and one or more ordinary microphones; the electronic device includes:
a processor;
a memory configured to store processor-executable instructions;
wherein the processor is configured to:
acquiring a plurality of paths of audio signals collected by the microphone array; the multi-channel audio signals comprise reference audio signals collected by the reference microphone and common audio signals collected by the common microphone;
determining a difference parameter between the normal audio signal and the reference audio signal based on a predetermined sample difference parameter;
correcting the common audio signal based on the difference parameter to obtain a corrected audio signal matched with the reference audio signal;
the processor is further configured to:
acquiring multi-path sample audio signals of a sample sound source collected by the microphone array; wherein the multi-path sample audio signal comprises a reference sample audio signal collected by the reference microphone and a common sample audio signal collected by the common microphone;
respectively determining sample difference parameters of each path of sample audio signal in the common sample audio signal and the reference sample audio signal;
the determining the sample difference parameters of each path of the common sample audio signal and the reference sample audio signal in the common sample audio signal respectively includes:
calculating a mean square value of the reference sample audio signal;
calculating a mean square value of the common sample audio signal;
calculating a cross-correlation function of the reference sample audio signal and each path of sample audio signal;
carrying out interpolation operation on the cross-correlation function to obtain an interpolation cross-correlation function;
and respectively determining the sample difference parameters of the sample audio signals and the reference sample audio signal according to the mean square value of the reference sample audio signal, the mean square value of the common sample audio signal and the interpolation cross-correlation function.
9. A computer-readable storage medium for processing audio signals acquired by a microphone array, the microphone array comprising a reference microphone and one or more ordinary microphones; the storage medium having stored thereon a computer program, the program when processed by a processor implementing:
acquiring a plurality of paths of audio signals collected by the microphone array; the multi-channel audio signals comprise reference audio signals collected by the reference microphone and common audio signals collected by the common microphone;
determining a difference parameter between the normal audio signal and the reference audio signal based on a predetermined sample difference parameter;
correcting the common audio signal based on the difference parameter to obtain a corrected audio signal matched with the reference audio signal;
the program when processed by the processor further implements:
acquiring multi-path sample audio signals of a sample sound source collected by the microphone array; wherein the multi-path sample audio signal comprises a reference sample audio signal collected by the reference microphone and a common sample audio signal collected by the common microphone;
respectively determining sample difference parameters of each path of sample audio signal in the common sample audio signal and the reference sample audio signal;
the determining the sample difference parameters of each path of the common sample audio signal and the reference sample audio signal in the common sample audio signal respectively includes:
calculating a mean square value of the reference sample audio signal;
calculating a mean square value of the common sample audio signal;
calculating a cross-correlation function of the reference sample audio signal and each path of sample audio signal;
carrying out interpolation operation on the cross-correlation function to obtain an interpolation cross-correlation function;
and respectively determining the sample difference parameters of the sample audio signals and the reference sample audio signal according to the mean square value of the reference sample audio signal, the mean square value of the common sample audio signal and the interpolation cross-correlation function.
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| WO2020215239A1 (en) * | 2019-04-24 | 2020-10-29 | 深圳市大疆创新科技有限公司 | Signal processing method and device for sound pickup apparatus, and computer storage medium |
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| CN111065036B (en) * | 2019-12-26 | 2021-08-31 | 北京声智科技有限公司 | Frequency response testing method and device of microphone array |
| CN112153547A (en) * | 2020-09-03 | 2020-12-29 | 海尔优家智能科技（北京）有限公司 | Audio signal correction method, audio signal correction device, storage medium and electronic device |
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