WO2015184893A1

WO2015184893A1 - Mobile terminal call voice noise reduction method and device

Info

Publication number: WO2015184893A1
Application number: PCT/CN2015/074770
Authority: WO
Inventors: 王进军; 孙焘; 薛华
Original assignee: 中兴通讯股份有限公司
Priority date: 2014-11-21
Filing date: 2015-03-20
Publication date: 2015-12-10
Also published as: WO2016078369A1; CN105611014A

Abstract

A mobile terminal call voice noise reduction method and device, the method comprising: data acquisition delays of two microphones of a mobile terminal are obtained; the relative positions between a sound source of a mobile terminal user and the two microphones are determined according to the delays; according to the relative positions, the microphone closer to the sound source is set as a main microphone, and the other microphone is set as an auxiliary microphone; on the basis of the main microphone and the auxiliary microphone, noise reduction processing is performed on a call voice of the mobile terminal via a noise reduction algorithm.

Description

Mobile terminal call voice noise reduction method and device

Technical field

This paper relates to the field of mobile communications, and in particular to a method and apparatus for voice noise reduction of a mobile terminal.

Background technique

At present, the application of mobile terminals has been very extensive and the popularity is very high. Voice call is a basic function of mobile terminals. The improvement of voice call quality is a topic that all mobile terminals are committed to.

Due to the uncertainty and complexity of the call environment, environmental noise is an important factor affecting the voice quality of the call. Therefore, the noise reduction performance of the mobile terminal has become a key indicator for the major operators to judge the mobile phone. How to eliminate or reduce the environmental noise has become Chip manufacturers and terminal manufacturers are committed to researching a key topic.

For mobile terminals, the noise reduction algorithm has evolved from previous single microphone (MIC) noise reduction to dual MIC noise reduction, and even microphone array algorithms. The number of MICs on mobile terminals has also evolved from a single MIC to a now-current dual MIC. Even some mobile terminals have already been 3MIC or 4MIC layouts. These new technologies are driving the development and performance of mobile terminal noise reduction technology. Upgrade.

In the current noise reduction algorithm, in order to achieve optimal noise reduction performance, the signal-to-noise ratio of the main MIC is required to be 6 dB larger than the signal-to-noise ratio of the sub-MIC. 1 is a schematic diagram of a dual MIC layout manner in the prior art. As shown in FIG. 2, due to the vocal characteristics of a conventional receiver, when the end user uses the handset mode to talk, the main MIC is closer to the mouth than the sub MIC, so It is easy to meet the requirements of the noise reduction algorithm. However, if the terminal user uses the hands-free mode to make a call, when the relative position of the person and the terminal is different, the signal-to-noise ratio of the primary and secondary MICs changes, so that the signal-to-noise ratio difference between the primary and secondary MICs cannot meet the requirements of the noise reduction algorithm. As a result, the noise reduction algorithm is invalidated, and the noise reduction performance of the call is drastically deteriorated. Therefore, a method is needed to identify such changes, and the noise reduction performance of the voice call is ensured, thereby improving the user experience and increasing the market competitiveness of the product.

Summary of the invention

Embodiments of the present invention provide a method and apparatus for voice noise reduction of a mobile terminal.

A mobile terminal call voice noise reduction method includes:

Obtaining a delay of data collected by two microphones of the mobile terminal;

Determining a relative position between the sound source of the mobile terminal user and the two microphones according to the delay;

According to the relative position, the microphone that is closer to the sound source is set as the primary microphone, and the other microphone is set as the secondary microphone;

Based on the primary microphone and the secondary microphone, the noise reduction algorithm is used to perform noise reduction processing on the call voice of the mobile terminal.

Optionally, the foregoing method further includes:

When the mobile terminal enters the call mode, it is determined by the voice path whether the call mode of the mobile terminal is a hands-free call mode, a hand-held call mode, or a headset call mode, and if it is determined to be a hand-held call mode or a headset call mode, the voice call is directly performed, if When it is determined that the hands-free call mode is activated, the primary and secondary microphone determination operations are started.

Optionally, the calculating the delay of data collected by the two microphones of the mobile terminal includes:

The collected data of the two microphone samples are respectively filtered by the band pass filter to remove the high frequency noise, and the filtered collected data is subjected to fast Fourier transform FFT to calculate two sets of speech spectra respectively corresponding to the two microphones;

Calculating power spectrum data of two sets of audio spectrums, and performing frequency domain weighting, performing inverse fast Fourier transform IFFT after the power spectrum data is accumulated to a predetermined number of frames, and obtaining an autocorrelation function;

Based on the calculated autocorrelation function, calculate the delay of the data collected by the two microphones according to Equation 1:

R12(τ)=E[x1(kT)x2(kT-τ)]=E[s(kT-τ1)s(kT-τ1-τ)]=Rss(τ-(τ1-τ2)) Equation 1;

Where τ corresponds to the maximum value of R12(τ) is the time delay τ12 between two microphones, x1(kT)=s(kT-τ1)+n1(kT), x2(kT)=s(kT-τ2 ) +n2(kT), s(kT) is the sound source signal, n1(kT) is the background noise of the first microphone, n2(kT) is the background noise of the second microphone, and τ1 and τ2 are the sound waves from the sound source to The propagation time of the first microphone and the second microphone, τ12=τ1-τ2 are two microphones The delay between the data collected, E[] represents the autocorrelation function between x1(kT) and x2(kT).

Optionally, determining, according to the calculated delay, the relative location between the sound source of the mobile terminal user and the two microphones includes:

Calculate the elevation angle of the sound source when the microphone array is used as the reference coordinate according to Formula 2.

Formula 2;

Where a is the distance between two microphones, τ is the delay of the data collected by the two microphones, c is the speed of sound, and d is the difference in sound path;

According to the elevation angle of the sound source

Determine the relative position between the source of the mobile terminal user and the two microphones.

Optionally, according to the determined relative position, the microphone that is closer to the sound source is set as the primary microphone, and the other microphone is set as the secondary microphone, including:

When the elevation angle of the sound source

When determining that the microphone at the bottom of the mobile terminal is closer to the sound source, when the elevation angle of the sound source

When it is determined that the microphone located at the top of the mobile terminal is closer to the sound source;

Set the microphone that is closer to the source to the primary microphone and the other microphone to the secondary microphone.

A mobile terminal call voice noise reduction device includes:

The obtaining module is configured to: obtain a delay of data collected by two microphones of the mobile terminal;

Determining a module, configured to: determine a relative position between the sound source of the mobile terminal user and the two microphones according to the delay;

Set the module, set to: set the microphone closer to the sound source as the primary microphone and the other microphone as the secondary microphone according to the relative position;

The noise reduction module is configured to: perform noise reduction processing on the call voice of the mobile terminal by using a noise reduction algorithm based on the primary microphone and the secondary microphone.

Optionally, the foregoing apparatus further includes:

The judging module is configured to: when the mobile terminal enters the call mode, determine, by using the voice path, whether the call mode of the mobile terminal is a hands-free call mode, a handheld call mode, or a headset call mode, such as If it is determined to be the handheld call mode or the headset call mode, the voice call is directly performed, and if it is determined to be the hands-free call mode, the primary and secondary microphone determination operations are started.

Optionally, the above obtaining module is set to:

Where τ corresponds to the maximum value of R12(τ) is the time delay τ12 between two microphones, x1(kT)=s(kT-τ1)+n1(kT), x2(kT)=s(kT-τ2 ) +n2(kT), s(kT) is the sound source signal, n1(kT) is the background noise of the first microphone, n2(kT) is the background noise of the second microphone, and τ1 and τ2 are the sound waves from the sound source to The propagation time of the first microphone and the second microphone, τ12=τ1-τ2 is the time delay of the data collected between the two microphones, and E[] represents the autocorrelation function between x1(kT) and x2(kT).

Optionally, the determining module is configured to:

Formula 2;

According to the elevation angle of the sound source

Optionally, the above setting module is set to:

When the elevation angle of the sound source

A computer readable storage medium storing program instructions that, when executed, implement the methods described above.

With the technical solution of the embodiment of the present invention, the primary and secondary MICs of the mobile terminal are dynamically adjusted according to the relative position changes of the user and the mobile terminal, and the noise reduction algorithm is implemented in the related art when the terminal user uses the hands-free mode for talking. The problem of failure can improve the sound quality of hands-free call reception voice without increasing the design cost and space of the mobile communication terminal. The noise reduction performance of the mobile terminal is maintained in a better state, thereby improving the voice quality of the call and improving the user experience.

BRIEF abstract

The drawings are only for the purpose of illustrating the embodiments, and are not intended to limit the invention. Throughout the drawings, the same reference numerals are used to refer to the same parts. In the drawing:

1 is a schematic diagram of a dual MIC layout manner in the related art;

2 is a flowchart of a method for reducing voice of a call of a mobile terminal according to an embodiment of the present invention;

3 is a basic basic principle diagram of a far field sound source localization according to an embodiment of the present invention;

4 is a schematic diagram of calculating a delay of data collected by two microphones according to an embodiment of the present invention;

Figure 5 is a flow chart showing the detailed processing of the embodiment of the present invention;

6 is a flow chart showing the judgment of the primary and secondary MICs according to an embodiment of the present invention;

FIG. 7 is a schematic structural diagram of a call voice noise reduction apparatus of a mobile terminal according to an embodiment of the present invention.

Embodiments of the invention

Exemplary embodiments of the present disclosure will be described in more detail below with reference to the accompanying drawings.

The invention provides a method and a device for reducing voice of a mobile terminal call voice, which are described below with reference to the accompanying drawings. Embodiments of the present invention will be described in detail. It is understood that the specific embodiments described herein are merely illustrative of the invention and are not intended to limit the invention.

Method embodiment

According to an embodiment of the present invention, a mobile terminal call voice noise reduction method is provided. FIG. 2 is a flowchart of a mobile terminal call voice noise reduction method according to an embodiment of the present invention. As shown in FIG. 2, according to an embodiment of the present invention, The mobile terminal call voice noise reduction method includes the following processing:

Step 201: Obtain a delay of data collected by two microphones of the mobile terminal.

In the actual application, before performing step 201, the following processing is first performed: when the mobile terminal enters the call mode, it is determined by the voice path whether the call mode of the mobile terminal is a hands-free call mode, a hand-held call mode, or a headset call mode, if it is determined In the handheld call mode or the headset call mode, the voice call is directly performed. If it is determined to be the hands-free call mode, the primary and secondary microphone determination operations are initiated (ie, steps 201-204 are performed).

In step 201, calculating the delay of data collected by the two microphones of the mobile terminal may include:

Step 1: respectively, the collected data of the two microphone samples are filtered by the band pass filter to remove the high frequency noise, and the filtered collected data is subjected to fast Fourier transform FFT to calculate two sets of voices respectively corresponding to the two microphones. Spectrum

Step 2: Calculate power spectrum data of two sets of audio spectrums, and perform frequency domain weighting, and perform inverse fast Fourier transform IFFT after the power spectrum data is accumulated to a predetermined number of frames, and obtain an autocorrelation function;

Step 3: Calculate the delay of the data collected by the two microphones according to Equation 1 based on the calculated autocorrelation function:

Step 202: Determine a relative position between the sound source of the mobile terminal user and the two microphones according to the delay; and include the following processing:

Step 1. Calculate the elevation angle of the sound source when the microphone array is used as the reference coordinate according to Formula 2.

Formula 2;

Step 2, according to the elevation angle of the sound source

Step 203, according to the relative position, the microphone that is closer to the sound source is set as the primary microphone, and the other microphone is set as the secondary microphone;

Optionally, in practical applications, when the elevation angle of the sound source

When it is determined, the microphone located at the bottom of the mobile terminal is closer to the sound source when the elevation angle of the sound source

It can be determined that the microphone located at the top of the mobile terminal is closer to the sound source;

Step 204: Perform noise reduction processing on the call voice of the mobile terminal by using a noise reduction algorithm based on the primary microphone and the secondary microphone.

The above technical solutions of the embodiments of the present invention are described in detail below with reference to the accompanying drawings.

3 is a basic basic principle diagram of a far-field sound source localization according to an embodiment of the present invention. First, as shown in FIG. 3, for a hands-free call, since the distance between the sound source and the mobile phone microphone is far, the mobile phone is The relative position of the microphone and the sound source can be regarded as the far field range, so the acoustic signal can be regarded as being transmitted in the form of a plane wave.

Suppose a is the distance between two MICs, τ is the delay of the signal reaching the two MICs, c is the speed of sound, and d is the difference of the sound path, then the elevation angle of the sound source when the microphone array is taken as the reference coordinate is obtained by the geometric relationship:

Therefore, by using a delay of τ, a horizontal angle can be obtained.

That is to say, the delay of the signal received by the two microphones can be used to calculate and determine the orientation of the sound source.

Assuming that the attenuation and reverberation of the sound intensity are not considered, the discrete models of the signals received by the two microphones in Figure 3 are:

X1(kT)=s(kT-τ1)+n1(kT), x2(kT)=s(kT-τ2)+n2(kT), where s(kT) is the sound source signal, n1(kT) And n2(kT) is the background noise. s(kT), n1(kT), and n2(kT) are not related to each other. Τ1 and τ2 are the propagation times of sound waves from the sound source to the microphone 1 and the microphone 2, and τ12=τ1-τ2 is the time delay between the two microphones.

The correlation function R12(τ) of x1(kT) and x2(kT) can be expressed as:

R12(τ)=E[x1(kT)x2(kT-τ)]=E[s(kT-τ1)s(kT-τ1-τ)]=Rss(τ-(τ1-τ2));

It can be seen from the nature of the autocorrelation function that when τ-(τ1-τ2) = 0, R12(τ) reaches a maximum value. Therefore, the τ corresponding to the maximum value of R12(τ) is the time delay τ12 between the two microphones. The elevation angle of the sound source, that is, the relative position between the sound source and the microphone, can be obtained from the time delay R12 between the two microphones.

Again, when the mobile terminal is in the hands-free call state, the delay between the two microphones of the mobile phone can be calculated by using the existing microphone array and the processing chip of the mobile phone.

4 is a schematic diagram of calculating a delay of data collected by two microphones according to an embodiment of the present invention. As shown in FIG. 4, the voice data sampled by the microphone is first filtered by a bandpass filter of 300 to 4 kHz to remove high frequency noise, and then The filtered speech signal is subjected to fast Fourier transform to obtain the speech spectrum; then the power spectrum data (cross-power spectrum) of the signal is obtained for the two sets of audio spectra obtained by the two microphones, and frequency domain weighting is performed, and the power spectrum data is accumulated. After reaching one frame, the inverse Fourier transform is used to find the autocorrelation function. Finally, the obtained autocorrelation function is used to obtain the delay of the data collected by the two microphones. Finally, the relative position between the sound source (speaker) and the microphone can be obtained according to the delay of the microphone, and the microphone closer to the sound source (speaker) is set as the main MIC, and the other MIC is set as the sub MIC, thereby Guaranteed noise reduction performance and improved hands-free calling sound quality.

FIG. 5 is a flowchart of detailed processing of an embodiment of the present invention. As shown in FIG. 5, the following specifically includes the following processing:

Step 501: The mobile terminal enters a call mode.

Step 502: The voice path is used to determine whether the mobile terminal is in the hands-free mode or the handheld or headset mode. If the call is in the handheld or headset mode, step 503 is performed to directly perform a voice call. If the voice is in the hands-free mode, step 504 is performed.

Step 503, directly performing a voice call;

Step 504, starting a primary and secondary MIC determination module;

Step 505, the process shown in FIG. 6 includes the following steps: Step 601, MIC1 and MIC2 receive a voice signal from a sound source (speaker); Step 602, using the above calculation method to calculate when the sound source reaches MIC1 and MIC2 Ττ12; Step 603, after obtaining the delay between MIC1 and MIC2, the formula can be utilized

Calculate the elevation angle between the sound source and MIC1 and MIC2

Finally, according to the elevation angle

The relative position between the sound source (speaker) and the MIC can be obtained when the elevation angle

When determining that the MIC1 in Figure 1 is closer to the sound source (speaker), when the elevation angle

At this time, the MIC2 distance sound source (speaker) in Fig. 1 is determined; in step 604, the main and sub MICs are determined.

506. According to the calculated relative position between the MIC1 and the MIC2, the MIC that is closer to the sound source (speaker) is set as the main MIC. If the main MIC setting of the original call is consistent with the calculated main MIC, the execution is performed. In step 507, if the main MIC setting of the original call does not match the calculated main MIC, steps 508 and 509 are performed.

Step 507, returning to the voice call;

Step 508, adjusting a noise reduction algorithm;

In step 509, a voice call is returned.

In the related art, the primary and secondary MIC information used in the hands-free call is constant, so when the relative position of the person and the terminal changes, the signal-to-noise ratio of the primary and secondary MIC may deteriorate, thereby affecting the noise reduction of the mobile terminal. Performance and voice quality of the call. The technical solution of the embodiment of the invention enables the user to update the primary and secondary MIC settings in real time when the position of the person changes with the relative position of the mobile terminal during the hands-free call, thereby keeping the noise reduction algorithm in a stable state and ensuring that the noise reduction algorithm is always in a stable state. The noise reduction performance of the mobile terminal compensates for the influence of the change of the position of the person on the performance of the noise reduction algorithm, thereby improving the sound quality.

Device embodiment

According to an embodiment of the present invention, a mobile terminal call voice noise reduction device is provided. FIG. 7 is a schematic structural diagram of a mobile terminal call voice noise reduction device according to an embodiment of the present invention. As shown in FIG. 7, according to an embodiment of the present invention, The mobile terminal call voice noise reduction device includes: an obtaining module 70, a determining module 72, a setting module 74, and a noise reduction module 76. The following is a detailed description of each module of the embodiment of the present invention. Detailed instructions.

The obtaining module 70 is configured to obtain a time delay of data collected by the two microphones of the mobile terminal; the obtaining module 70 is configured to:

The determining module 72 is configured to determine a relative position between the sound source of the mobile terminal user and the two microphones according to the calculated time delay; the determining module 72 may be configured to:

Formula 2;

According to the elevation angle of the sound source

The setting module 74 is configured to set the microphone closer to the sound source as the primary microphone and the other microphone as the secondary microphone according to the determined relative position; the setting module 74 may be configured as:

When the elevation angle of the sound source

The noise reduction module 76 is configured to perform noise reduction processing on the call voice of the mobile terminal by using a noise reduction algorithm based on the determined primary microphone and the secondary microphone.

In the embodiment of the present invention, the determining module further includes: when the mobile terminal enters the call mode, determining, by using the voice path, whether the call mode of the mobile terminal is a hands-free call mode, a handheld call mode, or a headset call mode, if it is determined to be handheld In the call mode or the headset call mode, the voice call is directly performed, and if it is determined to be the hands-free call mode, the primary and secondary microphone determination operations are started.

One of ordinary skill in the art will appreciate that all or a portion of the steps of the above-described embodiments can be implemented using a computer program flow, which can be stored in a computer readable storage medium, such as on a corresponding hardware platform (eg, The system, device, device, device, etc. are executed, and when executed, include one or a combination of the steps of the method embodiments.

Alternatively, all or part of the steps of the above embodiments may also be implemented by using an integrated circuit. These steps may be separately fabricated into individual integrated circuit modules, or multiple modules or steps may be fabricated into a single integrated circuit module. achieve.

The devices/function modules/functional units in the above embodiments may be implemented by a general-purpose computing device, which may be centralized on a single computing device or distributed over a network of multiple computing devices.

When each device/function module/functional unit in the above embodiment is implemented in the form of a software function module and sold or used as a stand-alone product, it can be stored in a computer readable storage medium. The above mentioned computer readable storage medium may be a read only memory, a magnetic disk or an optical disk or the like.

Industrial applicability

The embodiment of the invention keeps the noise reduction performance of the mobile terminal in a better state, thereby improving the voice quality of the call and improving the user experience.

Claims

A mobile terminal call voice noise reduction method includes:

Obtaining a delay of data collected by two microphones of the mobile terminal;

Determining, according to the time delay, a relative position between a sound source of the mobile terminal user and the two microphones;

According to the relative position, a microphone that is closer to the sound source is set as a primary microphone, and another microphone is set as a secondary microphone;

And performing noise reduction processing on the call voice of the mobile terminal by using a noise reduction algorithm based on the primary microphone and the secondary microphone.
The method of claim 1 further comprising:

When the mobile terminal enters the call mode, it is determined by the voice path that the call mode of the mobile terminal is a hands-free call mode, a handheld call mode, or a headset call mode, and if it is determined to be a handheld call mode or a headset call mode, directly The voice call, if it is determined to be the hands-free call mode, the primary and secondary microphone determination operations are initiated.
The method of claim 1, wherein calculating the delay of data collected by the two microphones of the mobile terminal comprises:

The collected data of the two microphone samples are respectively filtered by the band pass filter to remove the high frequency noise, and the filtered collected data is subjected to fast Fourier transform FFT to calculate two sets of speech spectra respectively corresponding to the two microphones;

Calculating power spectrum data of the two sets of audio spectrums, and performing frequency domain weighting, performing inverse fast Fourier transform IFFT after the power spectrum data is accumulated to a predetermined number of frames, and obtaining an autocorrelation function;

Based on the calculated autocorrelation function, calculate the delay of the data collected by the two microphones according to Equation 1:

R 12 (τ)=E[x 1 (kT)x 2 (kT-τ)]=E[s(kT-τ 1 )s(kT-τ 1 -τ)]=R ss (τ-(τ 1 -τ 2 )) Formula 1;

Where τ corresponds to the maximum value of R 12 (τ) is the time delay τ 12 between the two microphones, x 1 (kT)=s(kT-τ 1 )+n 1 (kT), x 2 (kT) =s(kT-τ 2 )+n 2 (kT), s(kT) is the sound source signal, n 1 (kT) is the background noise of the first microphone, and n 2 (kT) is the background noise of the second microphone. τ 1 and τ 2 are the propagation times of sound waves from the sound source to the first microphone and the second microphone, respectively, τ 12 = τ 1 - τ 2 is the time delay of the data collected between the two microphones, and E[] represents x 1 The autocorrelation function between (kT) and x 2 (kT).
The method of claim 3, wherein determining the relative position between the sound source of the mobile terminal user and the two microphones according to the calculated time delay comprises:

Calculating the elevation angle of the sound source when the microphone array is used as a reference coordinate according to Formula 2

Formula 2;

Where a is the distance between two microphones, τ is the delay of the data collected by the two microphones, c is the speed of sound, and d is the difference in sound path;

According to the elevation angle of the sound source
A relative position between the sound source of the mobile terminal user and the two microphones is determined.
The method of claim 4, wherein, according to the determined relative position, a microphone that is closer to the sound source is set as a primary microphone, and setting another microphone as a secondary microphone includes:

When the elevation angle of the sound source
When determining that the microphone at the bottom of the mobile terminal is closer to the sound source, when the elevation angle of the sound source
When it is determined that the microphone located at the top of the mobile terminal is closer to the sound source;

A microphone that is closer to the sound source is set as the primary microphone and another microphone is set as the secondary microphone.
A mobile terminal call voice noise reduction device includes:

The obtaining module is configured to: obtain a delay of data collected by two microphones of the mobile terminal;

a determining module, configured to: determine, according to the time delay, a relative position between a sound source of the mobile terminal user and the two microphones;

Setting a module, configured to: set a microphone closer to the sound source as a primary microphone and another microphone as a secondary microphone according to the relative position;

The noise reduction module is configured to: perform noise reduction processing on the call voice of the mobile terminal by using a noise reduction algorithm based on the primary microphone and the secondary microphone.
The apparatus of claim 6 further comprising:

The judging module is configured to: when the mobile terminal enters the call mode, determine, by using a voice path, whether the call mode of the mobile terminal is a hands-free call mode, a handheld call mode, or a headset call mode, if it is determined to be a handheld call mode or a headset In the call mode, the voice call is directly performed, and if it is determined to be the hands-free call mode, the primary and secondary microphone determination operations are started.
The apparatus of claim 5 wherein said obtaining module is configured to:

The collected data of the two microphone samples are respectively filtered by the band pass filter to remove the high frequency noise, and the filtered collected data is subjected to fast Fourier transform FFT to calculate two sets of speech spectra respectively corresponding to the two microphones;

Calculating power spectrum data of the two sets of audio spectrums, and performing frequency domain weighting, performing inverse fast Fourier transform IFFT after the power spectrum data is accumulated to a predetermined number of frames, and obtaining an autocorrelation function;

Based on the calculated autocorrelation function, calculate the delay of the data collected by the two microphones according to Equation 1:

R 12 (τ)=E[x 1 (kT)x 2 (kT-τ)]=E[s(kT-τ 1 )s(kT-τ 1 -τ)]=R ss (τ-(τ 1 -τ 2 )) Formula 1;

Where τ corresponds to the maximum value of R 12 (τ) is the time delay τ 12 between the two microphones, x 1 (kT)=s(kT-τ 1 )+n 1 (kT), x 2 (kT) =s(kT-τ 2 )+n 2 (kT), s(kT) is the sound source signal, n 1 (kT) is the background noise of the first microphone, and n 2 (kT) is the background noise of the second microphone. τ 1 and τ 2 are the propagation times of sound waves from the sound source to the first microphone and the second microphone, respectively, τ 12 = τ 1 - τ 2 is the time delay of the data collected between the two microphones, and E[] represents x 1 The autocorrelation function between (kT) and x 2 (kT).
The apparatus of claim 8 wherein said determining module is configured to:

Calculating the elevation angle of the sound source when the microphone array is used as a reference coordinate according to Formula 2

Formula 2;

Where a is the distance between two microphones, τ is the delay of the data collected by the two microphones, c is the speed of sound, and d is the difference in sound path;

According to the elevation angle of the sound source
A relative position between the sound source of the mobile terminal user and the two microphones is determined.
The apparatus of claim 9 wherein said setting module is configured to:

When the elevation angle of the sound source
When determining that the microphone at the bottom of the mobile terminal is closer to the sound source, when the elevation angle of the sound source
When it is determined that the microphone located at the top of the mobile terminal is closer to the sound source;

A microphone that is closer to the sound source is set as the primary microphone and another microphone is set as the secondary microphone.
A computer readable storage medium storing program instructions that, when executed, can implement the method of any of claims 1-5.