阅读说明：本技术 一种面向数字助听器的双麦克风语音增强方法和装置 (Double-microphone voice enhancement method and device for digital hearing aid ) 是由 熊志辉 陈旺 于 2021-08-25 设计创作，主要内容包括：本申请涉及一种面向数字助听器的双麦克风语音增强方法、装置、计算机设备和存储介质。所述方法包括：通过将双麦克风的语音信号变换到频域,得到第一频域信号和第二频域信号,将第二频域信号延时后得到第三频域信号,计算第一频域信号和第三频域信号的相位差后,根据相位差计算平滑系数,根据平滑系数将第一频域信号和第三频域信号进行融合,并设计滤波器,通过滤波器对混合频域信号进行滤波处理后,得到目标频域信号,再进行傅里叶反变换,得到输出的语音信号。本专利提出的方法,能够有效的提高双麦克风数字助听器输出的语音质量,且算法复杂度低,资源消耗较少,可以满足实时处理的需求。(The application relates to a dual-microphone speech enhancement method, a device, a computer device and a storage medium for a digital hearing aid. The method comprises the following steps: the method comprises the steps of converting voice signals of two microphones to a frequency domain to obtain a first frequency domain signal and a second frequency domain signal, delaying the second frequency domain signal to obtain a third frequency domain signal, calculating the phase difference between the first frequency domain signal and the third frequency domain signal, calculating a smoothing coefficient according to the phase difference, fusing the first frequency domain signal and the third frequency domain signal according to the smoothing coefficient, designing a filter, filtering the mixed frequency domain signal through the filter to obtain a target frequency domain signal, and performing Fourier inverse transformation to obtain an output voice signal. The method provided by the patent can effectively improve the voice quality output by the double-microphone digital hearing aid, has low algorithm complexity and less resource consumption, and can meet the requirement of real-time processing.)

1. A method for dual-microphone speech enhancement for a digital hearing aid, the method comprising:

acquiring a first voice signal of a first microphone and a second voice signal of a second microphone on the double-microphone digital hearing aid;

performing Fourier transform on the first voice signal and the second voice signal to obtain a first frequency domain signal and a second frequency domain signal, and performing time delay processing on the second frequency domain signal to obtain a third frequency domain signal;

obtaining a phase difference according to the first frequency domain signal and the third frequency domain signal, and obtaining a smooth coefficient according to a preset formula according to the phase difference; the preset formula comprises an adjusting factor; the adjusting factor is used for adjusting the importance degree of the third frequency domain signal;

weighting and summing the first frequency domain signal and the third frequency domain signal according to the smoothing coefficient to obtain a mixed frequency domain signal;

obtaining a filter according to the first frequency domain signal, the third frequency domain signal and the smoothing coefficient;

filtering the mixed frequency domain signal through the filter to obtain a target frequency domain signal;

and carrying out Fourier inverse transformation on the target frequency domain signal to obtain a voice signal output by the digital hearing aid.

2. The method of claim 1, wherein performing a fourier transform on the first speech signal and the second speech signal to obtain a first frequency-domain signal and a second frequency-domain signal, and performing a time-delay processing on the second frequency-domain signal to obtain a third frequency-domain signal comprises:

fourier transform is carried out on the first voice signal and the second voice signal to obtain a first frequency domain signal { X₁(t, ω) } and a second frequency-domain signal { X₂(t, ω) }, performing time delay processing on the second frequency domain signal to obtain a third frequency domain signal:

X₃(t,ω)＝e^-jωτX₂(t,ω)

where t denotes a frame number, ω denotes an angular frequency, { X₃(t, ω) } denotes the third frequency domain signal, j denotes an imaginary unit of complex numbers, and τ denotes a delay parameter of the second microphone.

3. The method of claim 2, wherein a phase difference is obtained from the first frequency domain signal and the third frequency domain signal, and a smoothing coefficient is obtained according to a preset formula according to the phase difference; the preset formula comprises an adjusting factor; the method comprises the following steps:

obtaining a phase difference according to the first frequency domain signal and the second frequency domain signal as follows:

wherein the content of the first and second substances,a phase operator is obtained;

obtaining a smoothing coefficient according to the phase difference and a preset formula as follows:

where φ (t, ω) is the smoothing coefficient, and a and b are adjustment factors.

4. The method of claim 3, wherein weighting and summing the first frequency-domain signal and the third frequency-domain signal according to the smoothing coefficient to obtain a mixed frequency-domain signal comprises:

weighting and summing the first frequency domain signal and the third frequency domain signal according to the smoothing coefficient to obtain a mixed frequency domain signal:

X₄(t,ω)＝φ(t,ω)·X₁(t,ω)+(1-φ(t,ω))·X₃(t,ω)

wherein, X₄(t, ω) is the mixed frequency domain signal.

5. The method of claim 3, wherein deriving a filter from the first frequency domain signal, the third frequency domain signal, and the smoothing coefficients comprises:

obtaining a filter according to the first frequency domain signal, the third frequency domain signal and the smoothing coefficient as follows:

where H (t, ω) is a filter.

6. The method of claim 5, wherein filtering the mixed frequency-domain signal through the filter to obtain a target frequency-domain signal comprises:

filtering the mixed frequency domain signal through the filter to obtain a target frequency domain signal:

X₅(t,ω)＝H(t,ω)*X₄(t,ω)

wherein, X₅(t, ω) is the target frequency domain signal.

7. The method according to any of claims 1 to 6, characterized in that the value of the adjustment factor is adjustable depending on the acquired parameters.

8. A dual microphone speech enhancement device for a digital hearing aid, the device comprising:

the signal acquisition module is used for acquiring a first voice signal of a first microphone and a second voice signal of a second microphone on the double-microphone digital hearing aid;

the signal transformation module is used for performing Fourier transformation on the first voice signal and the second voice signal to obtain a first frequency domain signal and a second frequency domain signal, and performing time delay processing on the second frequency domain signal to obtain a third frequency domain signal;

the smooth coefficient determining module is used for obtaining a phase difference according to the first frequency domain signal and the third frequency domain signal and obtaining a smooth coefficient according to a preset formula according to the phase difference; the preset formula comprises an adjusting factor; the adjusting factor is used for adjusting the importance degree of the third frequency domain signal;

a mixed frequency domain signal determining module, configured to perform weighted summation on the first frequency domain signal and the third frequency domain signal according to the smoothing coefficient to obtain a mixed frequency domain signal;

a filter determining module, configured to obtain a filter according to the first frequency domain signal, the third frequency domain signal, and the smoothing coefficient;

the target frequency domain signal determining module is used for filtering the mixed frequency domain signal through the filter to obtain a target frequency domain signal;

and the voice signal output module is used for carrying out Fourier inverse transformation on the target frequency domain signal to obtain a voice signal output by the digital hearing aid.

9. A computer device comprising a memory and a processor, the memory storing a computer program, wherein the processor implements the steps of the method of any one of claims 1 to 7 when executing the computer program.

10. A computer-readable storage medium, on which a computer program is stored, which, when being executed by a processor, carries out the steps of the method of any one of claims 1 to 7.

Technical Field

The present application relates to the field of hearing aids, and in particular, to a dual-microphone speech enhancement method and apparatus for a digital hearing aid, a computer device, and a storage medium.

Background

With the increasing form of aging of the global population and the continuous change of social environment, the number of hearing-impaired patients is increasing rapidly. Due to the limitations of the current medical level and condition, wearing digital hearing aids is currently the best choice for hearing-impaired patients.

In everyday life, however, the effectiveness of digital hearing aids is often affected by various types of interference noise. Research shows that under a steady-state noise environment, the hearing threshold of a hearing aid user is 10-15dB higher than that of a normal person; in the case of multiple persons speaking, 25dB or even higher is achieved. It is therefore of great importance for the hearing aid user to eliminate the ambient noise as much as possible.

Most of the traditional noise reduction methods are based on single-microphone voice noise reduction, mainly aiming at stable white noise, and the noise reduction effect is poor for non-stable noise such as human voice, music and automobiles.

Disclosure of Invention

In view of the above, there is a need to provide a method, an apparatus, a computer device and a storage medium for speech enhancement for digital hearing aids, which can have better noise reduction effect.

A dual microphone speech enhancement method for a digital hearing aid, the method comprising:

acquiring a first voice signal of a first microphone and a second voice signal of a second microphone on the double-microphone digital hearing aid;

performing Fourier transform on the first voice signal and the second voice signal to obtain a first frequency domain signal and a second frequency domain signal, and performing time delay processing on the second frequency domain signal to obtain a third frequency domain signal;

obtaining a phase difference according to the first frequency domain signal and the third frequency domain signal, and obtaining a smooth coefficient according to a preset formula according to the phase difference; the preset formula comprises an adjusting factor; the adjusting factor is used for adjusting the importance degree of the third frequency domain signal;

weighting and summing the first frequency domain signal and the third frequency domain signal according to the smoothing coefficient to obtain a mixed frequency domain signal;

obtaining a filter according to the first frequency domain signal, the third frequency domain signal and the smoothing coefficient;

filtering the mixed frequency domain signal through the filter to obtain a target frequency domain signal;

and carrying out Fourier inverse transformation on the target frequency domain signal to obtain a voice signal output by the digital hearing aid.

In one embodiment, the method further comprises the following steps: fourier transform is carried out on the first voice signal and the second voice signal to obtain a first frequency domain signal { X₁(t, ω) } and a second frequency-domain signal { X₂(t, ω) }, performing time delay processing on the second frequency domain signal to obtain a third frequency domain signal:

X₃(t,ω)＝e^-jωτX₂(t,ω)

where t denotes a frame number, ω denotes an angular frequency, { X₃(t, ω) } denotes the third frequency domain signal, j denotes an imaginary unit of complex numbers, and τ denotes a delay parameter of the second microphone.

In one embodiment, the method further comprises the following steps: obtaining a phase difference according to the first frequency domain signal and the second frequency domain signal as follows:

wherein the content of the first and second substances,a phase operator is obtained;

obtaining a smoothing coefficient according to the phase difference and a preset formula as follows:

where φ (t, ω) is the smoothing coefficient, and a and b are adjustment factors.

In one embodiment, the method further comprises the following steps: weighting and summing the first frequency domain signal and the third frequency domain signal according to the smoothing coefficient to obtain a mixed frequency domain signal:

X₄(t,ω)＝φ(t,ω)·X₁(t,ω)+(1-φ(t,ω))·X₃(t,ω)

wherein, X₄(t, ω) is the mixed frequency domain signal.

In one embodiment, the method further comprises the following steps: obtaining a filter according to the first frequency domain signal, the third frequency domain signal and the smoothing coefficient as follows:

where H (t, ω) is a filter.

In one embodiment, the method further comprises the following steps: filtering the mixed frequency domain signal through the filter to obtain a target frequency domain signal:

X₅(t,ω)＝H(t,ω)*X₄(t,ω)

wherein, X₅(t, ω) is the target frequency domain signal.

In one embodiment, the method further comprises the following steps: the value of the adjustment factor can be adjusted according to the acquired parameters.

A dual microphone speech enhancement device for a digital hearing aid, the device comprising:

the signal acquisition module is used for acquiring a first voice signal of a first microphone and a second voice signal of a second microphone on the double-microphone digital hearing aid;

the signal transformation module is used for performing Fourier transformation on the first voice signal and the second voice signal to obtain a first frequency domain signal and a second frequency domain signal, and performing time delay processing on the second frequency domain signal to obtain a third frequency domain signal;

the smooth coefficient determining module is used for obtaining a phase difference according to the first frequency domain signal and the third frequency domain signal and obtaining a smooth coefficient according to a preset formula according to the phase difference; the preset formula comprises an adjusting factor; the adjusting factor is used for adjusting the importance degree of the third frequency domain signal;

a mixed frequency domain signal determining module, configured to perform weighted summation on the first frequency domain signal and the third frequency domain signal according to the smoothing coefficient to obtain a mixed frequency domain signal;

a filter determining module, configured to obtain a filter according to the first frequency domain signal, the third frequency domain signal, and the smoothing coefficient;

the target frequency domain signal determining module is used for filtering the mixed frequency domain signal through the filter to obtain a target frequency domain signal;

and the voice signal output module is used for carrying out Fourier inverse transformation on the target frequency domain signal to obtain a voice signal output by the digital hearing aid.

A computer device comprising a memory and a processor, the memory storing a computer program, the processor implementing the following steps when executing the computer program:

acquiring a first voice signal of a first microphone and a second voice signal of a second microphone on the double-microphone digital hearing aid;

performing Fourier transform on the first voice signal and the second voice signal to obtain a first frequency domain signal and a second frequency domain signal, and performing time delay processing on the second frequency domain signal to obtain a third frequency domain signal;

obtaining a phase difference according to the first frequency domain signal and the third frequency domain signal, and obtaining a smooth coefficient according to a preset formula according to the phase difference; the preset formula comprises an adjusting factor; the adjusting factor is used for adjusting the importance degree of the third frequency domain signal;

weighting and summing the first frequency domain signal and the third frequency domain signal according to the smoothing coefficient to obtain a mixed frequency domain signal;

obtaining a filter according to the first frequency domain signal, the third frequency domain signal and the smoothing coefficient;

filtering the mixed frequency domain signal through the filter to obtain a target frequency domain signal;

and carrying out Fourier inverse transformation on the target frequency domain signal to obtain a voice signal output by the digital hearing aid.

A computer-readable storage medium, on which a computer program is stored which, when executed by a processor, carries out the steps of:

acquiring a first voice signal of a first microphone and a second voice signal of a second microphone on the double-microphone digital hearing aid;

performing Fourier transform on the first voice signal and the second voice signal to obtain a first frequency domain signal and a second frequency domain signal, and performing time delay processing on the second frequency domain signal to obtain a third frequency domain signal;

obtaining a phase difference according to the first frequency domain signal and the third frequency domain signal, and obtaining a smooth coefficient according to a preset formula according to the phase difference; the preset formula comprises an adjusting factor; the adjusting factor is used for adjusting the importance degree of the third frequency domain signal;

weighting and summing the first frequency domain signal and the third frequency domain signal according to the smoothing coefficient to obtain a mixed frequency domain signal;

obtaining a filter according to the first frequency domain signal, the third frequency domain signal and the smoothing coefficient;

filtering the mixed frequency domain signal through the filter to obtain a target frequency domain signal;

and carrying out Fourier inverse transformation on the target frequency domain signal to obtain a voice signal output by the digital hearing aid.

According to the double-microphone voice enhancement method, the double-microphone voice enhancement device, the computer equipment and the storage medium for the digital hearing aid, the voice signals of the double microphones are converted into the frequency domain to obtain the first frequency domain signal and the second frequency domain signal, the second frequency domain signal is delayed to obtain the third frequency domain signal, the phase difference between the first frequency domain signal and the third frequency domain signal is calculated, the smoothing coefficient is calculated according to the phase difference, the first frequency domain signal and the third frequency domain signal are fused according to the smoothing coefficient, the filter is designed, the mixed frequency domain signal is filtered through the filter to obtain the target frequency domain signal, and then the Fourier inversion is carried out to obtain the output voice signal. The method provided by the patent can effectively improve the voice quality output by the double-microphone digital hearing aid, has low algorithm complexity and less resource consumption, can meet the requirement of real-time processing, is very suitable for the application market of the hearing aid, and has very high practical value.

Drawings

FIG. 1 is a flow chart of a digital hearing aid-oriented dual-microphone speech enhancement method according to an embodiment;

FIG. 2 is an algorithm block diagram of a digital hearing aid oriented dual microphone speech enhancement method according to an embodiment;

FIG. 3 is a block diagram of a dual-microphone speech enhancement device for a digital hearing aid according to an embodiment;

FIG. 4 is a diagram illustrating an internal structure of a computer device according to an embodiment.

Detailed Description

In order to make the objects, technical solutions and advantages of the present application more apparent, the present application is described in further detail below with reference to the accompanying drawings and embodiments. It should be understood that the specific embodiments described herein are merely illustrative of the present application and are not intended to limit the present application.

The dual-microphone speech enhancement method for the digital hearing aid can be applied to the following application environments. The terminal executes a dual-microphone voice enhancement method facing a digital hearing aid, and obtains a first frequency domain signal and a second frequency domain signal by transforming voice signals of dual microphones to a frequency domain, obtains a third frequency domain signal by delaying the second frequency domain signal, calculates a phase difference between the first frequency domain signal and the third frequency domain signal, calculates a smooth coefficient according to the phase difference, fuses the first frequency domain signal and the third frequency domain signal according to the smooth coefficient, designs a filter, obtains a target frequency domain signal by filtering the mixed frequency domain signal through the filter, and obtains an output voice signal by performing Fourier inverse transformation. Wherein the terminal may be, but is not limited to, an embedded system in a digital hearing aid.

In one embodiment, as shown in fig. 1, there is provided a dual-microphone speech enhancement method for a digital hearing aid, comprising the steps of:

step 102, a first speech signal of a first microphone and a second speech signal of a second microphone of a dual-microphone digital hearing aid are acquired.

And 104, performing Fourier transform on the first voice signal and the second voice signal to obtain a first frequency domain signal and a second frequency domain signal, and performing time delay processing on the second frequency domain signal to obtain a third frequency domain signal.

The method of transforming a time domain signal into a frequency domain signal using fourier transform can help to know the characteristics of the signal from another perspective. The signal frequency spectrum represents the size of the component of the signal at different frequencies, and can provide more visual and rich information than the time domain signal waveform.

And 106, obtaining a phase difference according to the first frequency domain signal and the third frequency domain signal, and obtaining a smooth coefficient according to a preset formula according to the phase difference.

The preset formula comprises an adjusting factor, and the adjusting factor is used for adjusting the importance degree of the third frequency domain signal.

And 108, weighting and summing the first frequency domain signal and the third frequency domain signal according to the smooth coefficient to obtain a mixed frequency domain signal.

The mixing of the frequency domain signals is a weighted summation of the first frequency domain signal and the third frequency domain signal according to the smoothing coefficients.

And step 110, obtaining a filter according to the first frequency domain signal, the third frequency domain signal and the smoothing coefficient.

And 112, filtering the mixed frequency domain signal through a filter to obtain a target frequency domain signal.

And step 114, performing inverse Fourier transform on the target frequency domain signal to obtain a voice signal output by the digital hearing aid.

In the above two-microphone speech enhancement method for the digital hearing aid, as shown in fig. 2, the speech signals of the two microphones are converted to the frequency domain to obtain a first frequency domain signal and a second frequency domain signal, the second frequency domain signal is delayed to obtain a third frequency domain signal, the phase difference between the first frequency domain signal and the third frequency domain signal is calculated, a smoothing coefficient is calculated according to the phase difference, the first frequency domain signal and the third frequency domain signal are fused according to the smoothing coefficient, a filter is designed, the mixed frequency domain signal is filtered through the filter to obtain a target frequency domain signal, and then fourier inverse conversion is performed to obtain an output speech signal. The method provided by the patent can effectively improve the voice quality output by the double-microphone digital hearing aid, has low algorithm complexity and less resource consumption, can meet the requirement of real-time processing, is very suitable for the application market of the hearing aid, and has very high practical value.

In one embodiment, the method further comprises the following steps: fourier transform is carried out on the first voice signal and the second voice signal to obtain a first frequency domain signal { X₁(t, ω) } and a second frequency-domain signal { X₂(t, ω) }, performing time delay processing on the second frequency domain signal to obtain a third frequency domain signal:

X₃(t,ω)＝e^-jωτX₂(t,ω)

where t denotes a frame number, ω denotes an angular frequency, { X₃(t, ω) } denotes the third frequency domain signal, j denotes an imaginary unit of complex numbers, and τ denotes a delay parameter of the second microphone.

In one embodiment, the method further comprises the following steps: obtaining a phase difference according to the first frequency domain signal and the second frequency domain signal as follows:

wherein the content of the first and second substances,a phase operator is obtained;

obtaining a smoothing coefficient according to a preset formula according to the phase difference as follows:

where Φ (t, ω) is a smoothing coefficient, a and b are adjustment factors, and in this embodiment, a is 0.9 and b is 5.

In one embodiment, the method further comprises the following steps: weighting and summing the first frequency domain signal and the third frequency domain signal according to the smoothing coefficient to obtain a mixed frequency domain signal:

X₄(t,ω)＝φ(t,ω)·X₁(t,ω)+(1-φ(t,ω))·X₃(t,ω)

wherein, X₄(t, ω) is the mixed frequency domain signal.

In one embodiment, the method further comprises the following steps: the filter obtained according to the first frequency domain signal, the third frequency domain signal and the smoothing coefficient is:

where H (t, ω) is a filter.

In one embodiment, the method further comprises the following steps: filtering the mixed frequency domain signal through a filter to obtain a target frequency domain signal:

X₅(t,ω)＝H(t,ω)*X₄(t,ω)

wherein, X₅(t, ω) is the target frequency domain signal.

In one embodiment, the method further comprises the following steps: the value of the adjustment factor can be adjusted according to the acquired parameters.

It should be understood that, although the steps in the flowchart of fig. 1 are shown in order as indicated by the arrows, the steps are not necessarily performed in order as indicated by the arrows. The steps are not performed in the exact order shown and described, and may be performed in other orders, unless explicitly stated otherwise. Moreover, at least a portion of the steps in fig. 1 may include multiple sub-steps or multiple stages that are not necessarily performed at the same time, but may be performed at different times, and the order of performance of the sub-steps or stages is not necessarily sequential, but may be performed in turn or alternately with other steps or at least a portion of the sub-steps or stages of other steps.

In one embodiment, as shown in fig. 3, there is provided a dual-microphone speech enhancement device for a digital hearing aid, comprising: a signal acquisition module 302, a signal transformation module 304, a smoothing coefficient determination module 306, a mixed frequency domain signal determination module 308, a filter determination module 310, a target frequency domain signal determination module 312, and a speech signal output module 314, wherein:

a signal obtaining module 302, configured to obtain a first speech signal of a first microphone and a second speech signal of a second microphone of a dual-microphone digital hearing aid;

the signal transformation module 304 is configured to perform fourier transformation on the first voice signal and the second voice signal to obtain a first frequency domain signal and a second frequency domain signal, and perform delay processing on the second frequency domain signal to obtain a third frequency domain signal;

a smooth coefficient determining module 306, configured to obtain a phase difference according to the first frequency domain signal and the third frequency domain signal, and obtain a smooth coefficient according to a preset formula according to the phase difference; the preset formula comprises an adjusting factor; the adjusting factor is used for adjusting the importance degree of the third frequency domain signal;

a mixed frequency domain signal determining module 308, configured to perform weighted summation on the first frequency domain signal and the third frequency domain signal according to the smoothing coefficient to obtain a mixed frequency domain signal;

a filter determining module 310, configured to obtain a filter according to the first frequency domain signal, the third frequency domain signal, and the smoothing coefficient;

a target frequency domain signal determining module 312, configured to filter the mixed frequency domain signal through a filter to obtain a target frequency domain signal;

and the voice signal output module 314 is configured to perform inverse fourier transform on the target frequency domain signal to obtain a voice signal output by the digital hearing aid.

The signal transforming module 304 is further configured to perform fourier transform on the first voice signal and the second voice signal to obtain a first frequency domain signal { X₁(t, ω) } and a second frequency-domain signal { X₂(t, ω) }, performing time delay processing on the second frequency domain signal to obtain a third frequency domain signal:

X₃(t,ω)＝e^-jωτX₂(t,ω)

where t denotes a frame number, ω denotes an angular frequency, { X₃(t, ω) } denotes the third frequency domain signal, j denotes an imaginary unit of complex numbers, and τ denotes a delay parameter of the second microphone.

The smoothing coefficient determining module 306 is further configured to obtain a phase difference according to the first frequency domain signal and the second frequency domain signal as:

wherein the content of the first and second substances,a phase operator is obtained;

obtaining a smoothing coefficient according to a preset formula according to the phase difference as follows:

where φ (t, ω) is a smoothing coefficient, and a and b are adjustment factors.

The mixed frequency domain signal determining module 308 is further configured to perform weighted summation on the first frequency domain signal and the third frequency domain signal according to the smoothing coefficient to obtain a mixed frequency domain signal:

X₄(t,ω)＝φ(t,ω)·X₁(t,ω)+(1-φ(t,ω))·X₃(t,ω)

wherein, X₄(t, ω) is the mixed frequency domain signal.

The filter determining module 310 is further configured to obtain a filter from the first frequency domain signal, the third frequency domain signal and the smoothing coefficient as follows:

where H (t, ω) is a filter.

The target frequency domain signal determining module 312 is further configured to filter the mixed frequency domain signal through a filter, and obtain a target frequency domain signal as follows:

X₅(t,ω)＝H(t,ω)*X₄(t,ω)

wherein, X₅(t, ω) is the target frequency domain signal.

For specific definition of the dual-microphone speech enhancement device for digital hearing aids, reference may be made to the above definition of the dual-microphone speech enhancement method for digital hearing aids, which is not described herein again. The modules in the above-mentioned dual-microphone speech enhancement device for digital hearing aids may be implemented wholly or partly by software, hardware and a combination thereof. The modules can be embedded in a hardware form or independent from a processor in the computer device, and can also be stored in a memory in the computer device in a software form, so that the processor can call and execute operations corresponding to the modules.

In one embodiment, a computer device is provided, which may be a terminal, and its internal structure diagram may be as shown in fig. 4. The computer device includes a processor, a memory, a network interface, a display screen, and an input device connected by a system bus. Wherein the processor of the computer device is configured to provide computing and control capabilities. The memory of the computer device comprises a nonvolatile storage medium and an internal memory. The non-volatile storage medium stores an operating system and a computer program. The internal memory provides an environment for the operation of an operating system and computer programs in the non-volatile storage medium. The network interface of the computer device is used for communicating with an external terminal through a network connection. The computer program is executed by a processor to implement a dual microphone speech enhancement method for a digital hearing aid. The display screen of the computer equipment can be a liquid crystal display screen or an electronic ink display screen, and the input device of the computer equipment can be a touch layer covered on the display screen, and can also be a key, a track ball or a touch pad and the like arranged on the shell of the computer equipment.

Those skilled in the art will appreciate that the architecture shown in fig. 4 is merely a block diagram of some of the structures associated with the disclosed aspects and is not intended to limit the computing devices to which the disclosed aspects apply, as particular computing devices may include more or less components than those shown, or may combine certain components, or have a different arrangement of components.

In an embodiment, a computer device is provided, comprising a memory storing a computer program and a processor implementing the steps of the above method embodiments when executing the computer program.

In an embodiment, a computer-readable storage medium is provided, on which a computer program is stored, which computer program, when being executed by a processor, carries out the steps of the above-mentioned method embodiments.

It will be understood by those skilled in the art that all or part of the processes of the methods of the embodiments described above can be implemented by hardware instructions of a computer program, which can be stored in a non-volatile computer-readable storage medium, and when executed, can include the processes of the embodiments of the methods described above. Any reference to memory, storage, database, or other medium used in the embodiments provided herein may include non-volatile and/or volatile memory, among others. Non-volatile memory can include read-only memory (ROM), Programmable ROM (PROM), Electrically Programmable ROM (EPROM), Electrically Erasable Programmable ROM (EEPROM), or flash memory. Volatile memory can include Random Access Memory (RAM) or external cache memory. By way of illustration and not limitation, RAM is available in a variety of forms such as Static RAM (SRAM), Dynamic RAM (DRAM), Synchronous DRAM (SDRAM), Double Data Rate SDRAM (DDRSDRAM), Enhanced SDRAM (ESDRAM), Synchronous Link DRAM (SLDRAM), Rambus Direct RAM (RDRAM), direct bus dynamic RAM (DRDRAM), and memory bus dynamic RAM (RDRAM).

The technical features of the above embodiments can be arbitrarily combined, and for the sake of brevity, all possible combinations of the technical features in the above embodiments are not described, but should be considered as the scope of the present specification as long as there is no contradiction between the combinations of the technical features.

The above-mentioned embodiments only express several embodiments of the present application, and the description thereof is more specific and detailed, but not construed as limiting the scope of the invention. It should be noted that, for a person skilled in the art, several variations and modifications can be made without departing from the concept of the present application, which falls within the scope of protection of the present application. Therefore, the protection scope of the present patent shall be subject to the appended claims.