CN115209302A

CN115209302A - Audio noise reduction processing method, device, equipment, medium and program product

Info

Publication number: CN115209302A
Application number: CN202210975162.1A
Authority: CN
Inventors: 嵇守聪; 叶顺舟; 杨可欣
Original assignee: Spreadtrum Communications Shanghai Co Ltd
Current assignee: Spreadtrum Communications Shanghai Co Ltd
Priority date: 2022-08-12
Filing date: 2022-08-12
Publication date: 2022-10-18

Abstract

The application provides an audio noise reduction processing method, device, equipment, medium and program product. The method comprises the following steps: when the first audio signal is played by using a loudspeaker, acquiring a second audio signal acquired by an error microphone; processing the first audio signal by adopting an audio echo filter to obtain a third audio signal, wherein the coefficient of the audio echo filter is obtained by pre-design; compensating the third audio signal by using a compensation filter of the audio echo filter to obtain a fourth audio signal, wherein the compensation filter is used for compensating the audio echo filter so as to enable a target parameter of an error signal between the fourth audio signal and the retrieved first audio signal to be less than or equal to a preset threshold value; and eliminating the first audio signal acquired back in the second audio signal by using the fourth audio signal to obtain a fifth audio signal. The method can further eliminate the audio signals played by the loudspeaker and collected by the error microphone, and improves the noise reduction performance.

Description

Audio noise reduction processing method, device, equipment, medium and program product

Technical Field

The present application relates to the field of audio processing, and in particular, to an audio denoising processing method, apparatus, device, medium, and program product.

Background

In the process that a user uses the earphone, when the active noise reduction function of the earphone is turned on and the earphone is used for playing audio signals, the audio signals collected by the error microphone of the earphone comprise the audio signals played by the earphone. If no processing is done, the audio signal played by the earphone will be played again after being processed by the feedback filter, which will affect the sound quality.

The currently widely adopted solution is to add an audio echo filter to the earphone, which is essentially an audio echo filter that is designed in advance and has the same amplitude and frequency as the secondary channel and the opposite phase and frequency, so as to eliminate the audio signal played by the earphone and collected by the error microphone by using the audio echo filter.

However, in the actual use process of the earphone, after the audio echo filter is used to eliminate the audio signal collected by the error microphone, there may still be a residual audio signal collected by the error microphone and played by the earphone, which results in poor sound quality of the earphone.

Disclosure of Invention

The application provides an audio noise reduction processing method, device, equipment, medium and program product, which are used for improving the tone quality of an earphone.

In a first aspect, the present application provides an audio denoising processing method, including:

when the first audio signal is played by using the loudspeaker, a second audio signal collected by the error microphone is obtained; the second audio signal comprises: a first audio signal that is picked up passing through a target secondary channel between the speaker and the error microphone;

processing the first audio signal by adopting an audio echo filter to obtain a third audio signal; the coefficient of the audio echo filter is obtained by pre-design;

compensating the third audio signal by using a compensation filter of an audio echo filter to obtain a fourth audio signal, wherein the compensation filter is used for compensating the audio echo filter so as to enable a target parameter of an error signal of the fourth audio signal and the mined first audio signal to be less than or equal to a preset threshold value;

and eliminating the first audio signal acquired in the second audio signal by using the fourth audio signal to obtain a fifth audio signal.

Optionally, the second audio signal further includes: a noise signal of the internal environment, the method further comprising:

and processing the fifth audio signal by adopting a feedback filter to obtain a sixth audio signal.

Optionally, the method further includes:

and playing the sixth audio signal and the first audio signal by using the loudspeaker to eliminate the noise signal of the internal environment.

Optionally, the method further includes:

collecting a noise signal of an external environment by using a reference microphone;

processing the noise signal of the external environment by adopting a feedforward filter to obtain a seventh audio signal;

playing the sixth audio signal, the seventh audio signal, and the first audio signal with the speaker to cancel noise signals of an internal environment and an external environment.

Optionally, the method further includes:

playing a white noise signal x (n) with a speaker in response to the configuration instruction;

in the process of playing x (n) by using a loudspeaker, carrying out self-adaptive adjustment on the compensation filter according to an error signal between the audio signal compensated by the compensation filter and the audio signal collected by the error microphone to obtain a coefficient of the compensation filter;

configuring the compensation filter with the compensation filter coefficients.

Optionally, the target parameter is a mean square value, and in the process of playing x (n) by using a speaker, performing adaptive adjustment on the compensation filter according to an error signal between the audio signal compensated by the compensation filter and the audio signal collected by the error microphone to obtain the compensation filter coefficient includes:

acquiring an audio signal m (n) collected by an error microphone; the m (n) includes: x (n) mined through a target secondary channel between the speaker and the error microphone;

processing the x (n) by adopting an audio echo filter to obtain an audio signal y (n); the coefficient of the audio echo filter is obtained by pre-design;

compensating the y (n) by using a compensation filter of an audio echo filter to obtain an audio signal z (n);

acquiring an error signal e (n) between the z (n) and the m (n);

and utilizing the e (n) to perform self-adaptive adjustment on the coefficient of the compensation filter until the mean square value of the error signal e (n) is less than or equal to a preset mean square value threshold value, so as to obtain the coefficient of the compensation filter.

In a second aspect, the present application provides an audio noise reduction processing apparatus, the apparatus comprising:

the first acquisition module is used for acquiring a second audio signal acquired by the error microphone when the first audio signal is played by using the loudspeaker; the second audio signal comprises: a first audio signal that is picked up through a target secondary channel between the speaker and the error microphone;

the first processing module is used for processing the first audio signal by adopting an audio echo filter to obtain a third audio signal; the coefficient of the audio echo filter is obtained by pre-design;

the first compensation module is used for compensating the third audio signal by using a compensation filter of an audio echo filter to obtain a fourth audio signal, and the compensation filter is used for compensating the audio echo filter so as to enable a target parameter of an error signal between the fourth audio signal and the stoped first audio signal to be smaller than or equal to a preset threshold value;

and the elimination module is used for eliminating the first audio signal acquired back in the second audio signal by utilizing the fourth audio signal to obtain a fifth audio signal.

In a third aspect, the present application provides an audio noise reduction apparatus, comprising: a processor, and a memory communicatively coupled to the processor;

the memory stores computer-executable instructions;

the processor executes computer-executable instructions stored by the memory to implement the method of any of the first aspects.

In a fourth aspect, the present application provides a computer-readable storage medium having stored thereon computer-executable instructions for implementing the method according to any one of the first aspect when executed by a processor.

In a fifth aspect, the present application provides a computer program product comprising a computer program which, when executed by a processor, performs the method according to any one of the first aspect.

In a sixth aspect, the present application provides a chip having a computer program stored thereon, which, when executed by the chip, implements the method according to any one of the first aspects.

According to the audio noise reduction processing method, the audio noise reduction processing device, the audio noise reduction processing equipment, the audio noise reduction processing medium and the program product, the compensation filter is added to the audio echo filter, the inconsistency of the target secondary channel audio response and the audio echo filter is reduced, namely the audio echo filter is compensated through the compensation filter, and then the audio signal obtained after the audio signal is processed through the compensation filter can further offset the audio signal played by the loudspeaker picked by the error microphone, and the residual quantity of the picked audio signal is reduced. By the method, when the earphone is turned on to actively reduce noise and play the audio signal, the audio signal picked up by the error microphone played through the loudspeaker can be further reduced, and the noise reduction performance is improved while the tone quality is further improved.

Drawings

The accompanying drawings, which are incorporated in and constitute a part of this specification, illustrate embodiments consistent with the application and, together with the description, serve to explain the principles of the application.

Fig. 1 is a schematic structural diagram of an earphone provided in the prior art;

FIG. 2 is a schematic diagram of an audio denoising processing method provided in the prior art;

FIG. 3 is a schematic diagram of another audio denoising processing method provided by the prior art;

fig. 4 is a schematic diagram of an audio denoising processing method according to the present application;

FIG. 5 is a schematic diagram of another audio denoising method provided in the present application;

fig. 6 is a schematic structural diagram of an audio noise reduction processing apparatus provided in the present application;

fig. 7 is a schematic structural diagram of an audio noise reduction processing apparatus 700 according to the present application.

With the above figures, there are shown specific embodiments of the present application, which will be described in more detail below. These drawings and written description are not intended to limit the scope of the inventive concepts in any manner, but rather to illustrate the inventive concepts to those skilled in the art by reference to specific embodiments.

Detailed Description

Reference will now be made in detail to the exemplary embodiments, examples of which are illustrated in the accompanying drawings. When the following description refers to the accompanying drawings, like numbers in different drawings represent the same or similar elements unless otherwise indicated. The embodiments described in the following exemplary embodiments do not represent all embodiments consistent with the present application. Rather, they are merely examples of apparatus and methods consistent with certain aspects of the present application, as detailed in the appended claims.

The earphone referred to in the present application refers to an earphone having an active noise reduction function, which may be a wired earphone or a wireless earphone, and the present application does not limit the earphone. When it is a wireless earphone, it can be a bluetooth earphone, or a wifi earphone, etc.

Fig. 1 is a schematic structural diagram of a headset provided in the prior art, and as shown in fig. 1, the headset includes: error microphone, reference microphone, speaker, the processing chip who sets up in the earphone.

The error microphone is used for acquiring a noise signal of an internal environment. The internal environment refers to a closed environment formed by the earphone and the inner ear canal when the earphone is used.

And the reference microphone is used for acquiring a noise signal of the external environment. The external environment is an environment other than a closed environment formed by the earphone and the user's internal auditory canal when the earphone is used. It should be understood that the external environment is relative to the internal environment.

The processing chip refers to a chip which is located inside the earphone and used for processing audio signals, a feedback filter and a feedforward filter are arranged in the processing chip, the processing chip is used for achieving the noise reduction function of the earphone, and the specific implementation mode is as follows:

fig. 2 is a schematic diagram of an audio noise reduction processing method provided in the prior art, as shown in fig. 2, a (z) is a feedforward filter, and C (z) is a feedback filter. The illustrated audio signal output module may be a device or a component connected to the earphone, such as a mobile phone, a tablet, and the like, which is not limited in this application.

The error microphone collects the noise signal (3) of the internal environment and outputs the noise signal to a feedback filter C (z) of the processing chip for processing, and outputs an audio signal (4) to the loudspeaker.

The feedforward filter is essentially a filter coefficient configured in the processing chip for processing the noise signal of the internal environment collected by the error microphone, and is not a hardware entity. The noise signal of the internal environment can be processed through a feedforward filter, and an audio signal with the same amplitude and the opposite phase of the noise signal of the internal environment is obtained. Because the amplitude is the same, the audio signals with opposite phases can be mutually cancelled, therefore, the loudspeaker plays the audio signal (4) obtained by processing, and the noise signal of the internal environment can be eliminated.

The reference microphone collects noise signals (1) of the external environment and then transmits the noise signals to a feedforward filter A (z) of the processing chip. After being processed by the feedforward filter a (z), the audio signal (2) is output to a loudspeaker.

Similar to the feedforward filter, the feedback filter is essentially a filter coefficient configured in the processing chip for processing the noise signal of the external environment collected by the reference microphone, and is not a hardware entity. The noise signal of the external environment can be processed through the feedback filter, and the audio signal with the same amplitude and the opposite phase as the noise signal actually heard by the human ear when the noise signal of the external environment is transmitted to the human ear is obtained. The loudspeaker plays the audio signal (2) obtained by processing, and can eliminate the noise signal of the external environment.

By the mode, the noise signals in the internal environment and the external environment can be eliminated while the audio (5) output by the audio output module is played, namely, the noise reduction effect is achieved.

However, when the earphone is turned on for both active noise reduction and audio signal playing, the error microphone collects the noise signal of the internal environment and also collects the audio signal output by the audio output module and played through the speaker, and the audio signal is processed by the feedback filter C (z) and then output to the speaker again for playing, which affects the sound quality of the earphone.

Fig. 3 is a schematic diagram of another audio noise reduction processing method provided by the prior art, and as shown in fig. 3, at present, an audio echo filter M (z) with the same amplitude and frequency and the opposite phase and frequency as the secondary channel is added in a processing chip in advance to solve the above problem.

Specifically, before a noise signal (3) collected by an error microphone and including an audio signal played through a loudspeaker is processed by a feedback filter C (z), an echo filter M (z) processes an audio signal (5) to obtain an audio signal (6).

The secondary channel contains the acoustic path between the loudspeaker and the error microphone and may also include one or more of a digital-to-analog converter (DAC), a reconstruction filter, a power amplifier, a pre-amplification and anti-aliasing filter, and an analog-to-digital converter (ADC), depending on the actual configuration of the headset. The audio echo filter obtained by the pre-design can be an audio echo filter which is obtained by utilizing artificial ear simulation to be actually used under the condition that the earphone is delivered before the factory and has the same amplitude and frequency as the secondary channel and opposite phase and frequency; or an audio echo filter which is consistent with the amplitude frequency and opposite to the phase frequency of the secondary channel can be obtained by a line down-sampling mode when the earphone is actually used by a test sample user before leaving a factory. The present application does not limit the preset acquisition mode of the audio echo filter.

Because the amplitude and frequency of the audio echo filter M (z) are consistent with those of the secondary channel and the phase and frequency of the audio echo filter M (z) are opposite, the audio signal (6) obtained by processing through the audio echo filter and the audio signal (3) collected by the error microphone can be combined in the processing chip, so that the collected audio signal which is identical to the audio signal (6) in amplitude and opposite in phase in the audio signal (3) can be offset, and an audio signal (7) is obtained. Thus, the audio signal (7) is processed by the feedback processor C (z) and then transmitted to the loudspeaker, so that the audio signal collected by the error microphone can be reduced and played by the loudspeaker.

However, when the earphone is actually used, there still may be a part of residual audio signals collected by the error microphone and not completely processed, which are played again through the speaker, and the sound quality in actual use is affected.

The inventor researches and discovers that the structure of an acoustic device of the earphone is often inconsistent, and in addition, due to individual wearing differences such as different sizes and fitting degrees of ear canals, a preset audio echo filter can not achieve the purposes that the amplitude and the frequency of a secondary channel are consistent and the phase and frequency of the secondary channel are opposite when the earphone is used currently, so that after the audio signal acquired by an error microphone is subjected to offset processing, more music signal residues are left before the audio signal enters a feedback filter, and the influence is generated on the tone quality.

In view of the above, the present application provides an audio denoising processing method. According to the method, the preset audio echo filter M (z) is connected with the compensation filter W (z) in a cascade mode, the preset audio echo filter M (z) is consistent with the amplitude and the frequency of a secondary channel of the earphone in actual use after leaving a factory, and the phase frequency of the secondary channel is opposite, so that the audio signal collected by the error microphone is further eliminated before entering the compensation filter, and the tone quality effect is improved.

The execution main body of the application can be a headset, or a processing chip in the headset. The following describes the technical solutions of the present application and how to solve the above technical problems with specific embodiments. These several specific embodiments may be combined with each other below, and details of the same or similar concepts or processes may not be repeated in some embodiments. Embodiments of the present application will be described below with reference to the accompanying drawings.

Fig. 4 is a schematic diagram of an audio denoising processing method provided in the present application, and as shown in fig. 4, the audio denoising processing method provided in the present application includes the following steps:

s101, when the first audio signal is played by the loudspeaker, a second audio signal collected by the error microphone is obtained.

The first audio signal, i.e., the graphic audio signal (5), refers to an audio signal played by the earphone during operation and provided for the user to listen to, such as a music signal, a voice call audio signal, a short video audio signal, and the like.

The second audio signal includes: the first audio signal, i.e. the graphical audio signal (3), is picked up through the target secondary channel between the loudspeaker and the error microphone.

The target secondary channel refers to a secondary channel when the user actually uses the earphone after leaving a factory.

After passing through the target secondary channel, the first audio signal (5) is subjected to a change in amplitude and phase by the target secondary channel. This results in a difference in phase and amplitude between the first audio signal picked up by the error microphone and the first audio signal (5) before being played with the loudspeaker.

S102, processing the first audio signal by adopting an audio echo filter to obtain a third audio signal.

The coefficients of the audio echo filter are pre-designed.

As described above, the preset coefficient of the audio echo filter may be the coefficient of the audio echo filter that is obtained by using artificial ear simulation before the earphone leaves the factory and has the same amplitude and frequency as the secondary channel and opposite phase and frequency; or by means of line down sampling, when the test sample is actually used by a user, an audio echo filter with the amplitude and frequency consistent with those of the secondary channel and the phase frequency opposite to those of the secondary channel can be obtained. The method for obtaining the coefficient of the preset audio echo filter is not limited in the present application.

Due to the structural inconsistency of the acoustic devices and the individual wearing difference, the preset audio echo filter may not be consistent with the amplitude and frequency and opposite to the phase and frequency of the target secondary channel, and therefore if the third audio signal (6) obtained after being processed by the audio echo filter directly processes the second audio signal (3) collected by the error microphone, more audio signal residues played by the collected loudspeaker exist.

And S103, compensating the third audio signal by using a compensation filter of the audio echo filter to obtain a fourth audio signal.

The compensation filter is essentially a filter coefficient in a processing chip for processing an audio signal obtained after processing by the audio echo filter, and is not a hardware entity. Similar to the aforementioned feedforward filter, the compensation filter coefficients are used to cancel the audio signal picked up by the error microphone.

In this step, the audio echo filter M (z) is compensated by using a compensation filter W (z) cascaded with the audio echo filter M (z), that is, the coefficient of the audio echo filter is compensated to compensate for the third audio signal (6) obtained after being processed by the audio echo filter M (z) to obtain the fourth audio signal (9), so that the target parameter of the error signal between the fourth audio signal (9) and the retrieved first audio signal is less than or equal to the preset threshold. The fourth audio signal (9) is in phase opposition to the audio signal picked up by the error microphone.

The target parameter is used for representing the elimination degree of the audio signal played by the loudspeaker picked up by the error microphone, and the smaller the target parameter is, the more thoroughly the error signal is eliminated. The preset threshold value can be set according to actual requirements, and is not limited in the application.

And S104, eliminating the first audio signal which is back-collected in the second audio signal by using the fourth audio signal to obtain a fifth audio signal.

The elimination is achieved in the processing chip by adding the fourth audio signal (9) and the second audio signal (3). Because the audio signals with the same amplitude and opposite phases can be mutually offset, the first audio signal acquired back in the second audio signal can be eliminated by using the fourth audio signal in the above way.

Compared with the third audio signal (6) obtained after being processed by the audio echo filter M (z), the fourth audio signal (9) obtained after being processed by the compensation filter W (z) can further eliminate the first audio signal acquired by the error microphone, and reduce the residue of the first audio signal (5) acquired in the fifth audio signal (r) finally obtained.

In this embodiment, by adding the compensation filter to the audio echo filter, it is further achieved that the amplitude and the frequency of the target secondary channel audio and the amplitude and the frequency of the audio echo filter are consistent and the phases of the target secondary channel audio and the audio echo filter are opposite, that is, the audio echo filter is compensated by the compensation filter, so that the audio signal obtained after the processing of the compensation filter can further cancel the audio signal played by the speaker picked by the error microphone, and the residual amount of the picked audio signal is reduced. By the method, when the earphone is turned on to actively reduce noise and play the audio signal, the audio signal collected by the error microphone played through the loudspeaker can be further reduced, and the noise reduction performance is improved while the tone quality is further improved.

With reference to fig. 4, if the earphone employs an error microphone to eliminate the noise signal of the internal environment to realize the active noise reduction function, the second audio signal may include the noise signal of the internal environment in addition to the first audio signal that is extracted, and after step S104 in the above embodiment, the following steps may be further included:

and S201, processing the fifth audio signal by using a feedback filter to obtain a sixth audio signal.

Since the second audio signal (3) comprises a noise signal of the internal environment, the fifth audio signal also comprises a noise signal of the internal environment.

In this step, a feedback filter C (z) is used to reverse phase fifth audio signal (R) and, in turn, a sixth audio signal (R) having the same amplitude and opposite phase as fifth audio signal (R) is obtained

S202, playing the sixth audio signal and the first audio signal by using a loudspeaker to eliminate the noise signal of the internal environment.

In this step, the sixth audio signal is used

Is an audio signal of the same amplitude and opposite phase as fifth audio signal (R), which includes noise signals of the internal environment, so that sixth audio signal (R) is played using the loudspeaker

And after the first audio signal (5), the noise signal of the internal environment can be eliminated while the first audio signal (5) is played, and the noise reduction function of the earphone is realized.

With reference to fig. 4, if the earphone employs an error microphone and a reference microphone, and the active noise reduction function is implemented by eliminating the noise signal of the internal environment and the noise signal of the external environment, the method may further include the following steps:

and S301, collecting a noise signal of an external environment by using a reference microphone.

This step is to use the reference microphone to realize the collection of the noise signal (1) in the external environment as shown above.

S302, processing the noise signal of the external environment by adopting a feedforward filter to obtain a seventh audio signal.

And (3) inverting the noise signal (1) of the external environment through a feedforward filter A (z) to obtain a seventh audio signal (2) which has the same amplitude and opposite phase with the noise signal (1) of the external environment.

And S303, playing the sixth audio signal by using a loudspeaker to eliminate the noise signals of the internal environment and the external environment.

In the step, the seventh audio signal (2) is an audio signal which has the same amplitude and opposite phase with the noise signal (1) of the external environment. Thus playing the sixth audio signal

The seventh audio signal (2) and the first audio signal (5) can realize that the noise signals transmitted to human ears by the internal environment and the external environment can be eliminated while the first audio signal (5) is played.

In the audio noise reduction processing method provided by the embodiment, the noise signal transmitted from the external environment to the human ear is reduced by the feedforward filter, and the noise signal in the internal environment is reduced by the feedback filter. Like this when eliminating the music signal of poor microphone back production, can also eliminate the noise signal that external environment transmitted to the people's ear and in the internal environment, further promoted the noise reduction of earphone, guaranteed music tone quality, promote user experience.

It should be noted that, in the foregoing embodiments, parts that are the same as those in the prior art are not described again, and the implementation manner thereof is similar. For example, a feedforward filter a (z) may be used to process noise signal (1) in the external environment, and a feedback filter C (z) may be used to process fifth audio signal (r).

The compensation filter related in the above embodiment of the audio noise reduction method may be configured offline before the earphone leaves the factory, or configured by itself based on the secondary channel when the user actually uses the compensation filter. How the compensation filter is configured is explained below.

Fig. 5 is a schematic diagram of another audio noise reduction method provided in the present application, as shown in fig. 5, for example, the method may include the following steps:

s401, responding to the configuration instruction, and playing a white noise signal x (n) by using a loudspeaker.

The configuration instructions may be triggered by software installed in the device to which the headset is connected. The device may be, for example, a mobile phone, a computer, a tablet, and the like, which is not limited in this application. The connection means may be a wired or wireless connection. For example, by clicking a corresponding key in the software, a configuration instruction may be sent to the headset through the device. Alternatively, the user may trigger the configuration instruction by tapping the headset (e.g., tapping the headset 3 times in a row). The present application does not limit the manner in which the configuration instruction is triggered.

The white noise signal x (n) may be stored in the earphone or obtained from a control device connected to the earphone, which is not limited in the present application.

As a possible implementation manner, after the headset acquires the configuration instruction sent by the device connected to the headset, the white noise signal x (n) is acquired from the device, and then the white noise signal x (n) is played by using the speaker.

S402, in the process of playing x (n) by using a loudspeaker, carrying out self-adaptive adjustment on the compensation filter according to an error signal between the audio signal compensated by the compensation filter and the audio signal collected by the error microphone to obtain a compensation filter coefficient.

The above adaptive adjustment may be implemented by an adaptive algorithm. The adaptive algorithm may be any one of a Recursive Least Square (RLS) algorithm, a Least Mean Square (LMS) algorithm, a Normalized Least Mean Square (LMS) algorithm, a Variable Least Mean Square (VLMS) algorithm, and the like. The adjustment method is not limited in the present application, and those skilled in the art can set the adjustment method according to actual requirements.

With continued reference to fig. 5, in this step, the error signal e (n), i.e. the audio signal z (n) compensated by the compensation filter W (z), is cancelled out with the audio signal m (n) collected by the error microphone. The difference between the compensated audio signal z (n) and the retrieved audio signal m (n) can be made clear by the error signal e (n), and then the compensation filter is adaptively adjusted based on the error signal, so that the fitting degree of the compensated audio signal and the retrieved audio signal is improved, the residual error signal is reduced, and the coefficient of the compensation filter can be obtained until the target parameter of the error signal e (n) is smaller than the preset threshold value.

The target parameter, that is, the parameter for determining the difference between the compensated audio signal z (n) and the recovered audio signal m (n), is specifically determined according to the adaptive adjustment method used, and for example, when the LMS algorithm is used, the target parameter may be a mean square value. This application is not limited thereto.

And S403, configuring a compensation filter by using the compensation filter coefficient.

In this step, the obtained compensation filter coefficient is determined to be the coefficient of the compensation filter for compensating the third audio signal in the above embodiment.

In this embodiment, the audio signal m (n) picked up by the microphone is an audio signal transmitted through a stimulation channel during actual use, and the difference between the two audio signals during actual use can be reflected by the compensated audio signal z (n) and the error signal e (n) obtained after the error microphone picked up audio signal m (n) is cancelled. And carrying out self-adaptive adjustment on the compensation filter through the error signal e (n), and obtaining the coefficient of the compensation filter when the target parameter of the error signal e (n) is smaller than a preset threshold value.

The method can determine the compensation filter coefficient according to the secondary channel frequency response in actual use, can make up for the fact that the secondary channel frequency response cannot be consistent with the amplitude frequency and opposite to the phase frequency of the preset audio echo filter in actual use, further more accurately eliminates the audio signal acquired by the error microphone, and improves the tone quality.

In addition, the configuration process can be completed offline before the earphone is produced, and can also be completed according to the actual use scene and the secondary channel configuration when the user uses the earphone, so that the diversified use requirements of the user can be met, and the noise can be further eliminated.

In the following, taking the LMS adaptive algorithm as an example, how to adaptively adjust the compensation filter according to the error signal between the audio signal compensated by the compensation filter and the audio signal collected by the error microphone in the process of playing x (n) by using the speaker, so as to obtain the compensation filter coefficient, that is, step S402 in the above embodiment is described. The same or similar contents in this embodiment as those in the above embodiments can refer to the above embodiments, and are not described herein again. With continued reference to fig. 5, the above step S402 includes:

s501, obtaining an audio signal m (n) collected by an error microphone.

The above m (n) includes: x (n) is extracted through the target secondary channel between the loudspeaker and the error microphone.

And S502, processing x (n) by adopting an audio echo filter to obtain an audio signal y (n).

The audio echo filter is in accordance with the frequency response of the simulated secondary channel.

And S503, compensating y (n) by using a compensation filter of the audio echo filter to obtain an audio signal z (n).

In this step, during the iterative process of performing adaptive adjustment on the compensation filter, the compensation filter uses the initial compensation filter coefficient to compensate y (n) during the first iterative cycle. The initial compensation filter coefficient may be a default value set when the headphone leaves the factory, or may be a compensation filter coefficient value configured after the compensation filter coefficient is adjusted last time, which is not limited in the present application.

S504, acquiring an error signal e (n) between z (n) and m (n).

And the signal obtained after z (n) and m (n) are cancelled is the error signal e (n).

And S505, judging whether the mean square value of the error signal e (n) is less than or equal to a preset mean square value threshold value.

If yes, ending the process to obtain the compensation filter coefficient. That is, the compensation filter coefficients employed in this iteration loop are taken as the determined compensation filter coefficients.

If not, go to step S506.

And S506, adjusting the coefficient of the compensation filter.

After the coefficients of the compensation filter are adjusted, the process returns to step S503.

Namely, the coefficient of the compensation filter is adaptively adjusted by using e (n) until the mean square value of the error signal e (n) is less than or equal to the preset mean square value threshold.

Specifically, the compensation filter coefficients may be adjusted with reference to the following formula.

y(n)＝x(n)M ^T (n)

z(n)＝y(n)W ^T (n)

W(n+1)＝W(n)+2μe(n)x(n)

Where n denotes the nth time, and the superscript T denotes transposition. The third formula is a newer formula of the standard time-domain LMS algorithm. The above equation is updated point by point, i.e. the filter coefficients are updated each time a new x (n) and e (n) is given, where μ denotes the step size of the update.

In this embodiment, the compensation filter coefficient is obtained by the LMS adaptive algorithm, and it is further possible to achieve that the amplitude and frequency of the audio echo filter M (z) are the same as and opposite to the phase and frequency of the secondary channel in actual use, so that the audio signal collected by the error microphone is further eliminated before entering the compensation filter, and the sound quality of the earphone is improved.

Fig. 6 is a schematic structural diagram of an audio noise reduction processing apparatus provided in the present application. As shown in fig. 6, the audio noise reduction processing apparatus includes: the first obtaining module 11, the processing module 12, the compensation module 13, and the cancellation module 14, optionally, the audio noise reduction processing apparatus may include at least one of the following modules: the device comprises a playing module 15, a compensation adjusting module 16, a configuration module 17 and a second obtaining module 18.

The first obtaining module 11 is configured to obtain a second audio signal collected by the error microphone when the first audio signal is played by using the speaker; the second audio signal comprises: a first audio signal that is picked up through a target secondary channel between the speaker and the error microphone;

a processing module 12, configured to process the first audio signal by using an audio echo filter to obtain a third audio signal; the coefficient of the audio echo filter is obtained by pre-design;

a compensation module 13, configured to compensate the third audio signal by using a compensation filter of an audio echo filter to obtain a fourth audio signal, where the compensation filter is configured to compensate the audio echo filter, so that a target parameter of an error signal between the fourth audio signal and the mined first audio signal is less than or equal to a preset threshold;

and the eliminating module 14 is configured to eliminate the first audio signal acquired in the second audio signal by using the fourth audio signal, so as to obtain a fifth audio signal.

As a possible implementation, the second audio signal further includes: noise signals of the internal environment. The processing module 12 is further configured to process the fifth audio signal by using a feedback filter to obtain a sixth audio signal.

In this implementation, the playing module 15 is configured to play the sixth audio signal and the first audio signal through the speaker to eliminate the noise signal of the internal environment.

Or, the first obtaining module 11 is further configured to collect a noise signal of an external environment by using a reference microphone; the processing module 12 is further configured to process the noise signal of the external environment by using a feedforward filter to obtain a seventh audio signal; a playing module 15, configured to play the sixth audio signal, the seventh audio signal, and the first audio signal with the speaker to eliminate noise signals of an internal environment and an external environment.

As a possible implementation manner, the playing module 15 is configured to play the white noise signal x (n) with a speaker in response to the configuration instruction; the compensation adjusting module 16 is configured to perform adaptive adjustment on the compensation filter according to an error signal between the audio signal compensated by the compensation filter and the audio signal collected by the error microphone in the process of playing x (n) by using the speaker, so as to obtain a coefficient of the compensation filter; a configuration module 17, further configured to configure the compensation filter using the compensation filter coefficients.

As a possible implementation, the target parameter is a mean square value. The first obtaining module 11 is further configured to obtain an audio signal m (n) collected by the error microphone; the m (n) includes: x (n) mined through a target secondary channel between the loudspeaker and the error microphone; the coefficients of the audio echo filter are pre-designed.

The processing module 12 is further configured to process the x (n) by using an audio echo filter to obtain an audio signal y (n);

the compensation module 13 is further configured to compensate y (n) by using a compensation filter of the audio echo filter, so as to obtain an audio signal z (n);

a second obtaining module 18, configured to obtain an error signal e (n) between z (n) and m (n);

the compensation adjustment module 16 is specifically configured to perform adaptive adjustment on the coefficient of the compensation filter by using the e (n) until the mean square value of the error signal e (n) is less than or equal to a preset mean square value threshold, so as to obtain the coefficient of the compensation filter.

The audio noise reduction processing apparatus provided in the embodiment of the present application may execute the audio noise reduction processing method in the foregoing method embodiment, and the implementation principle and the technical effect are similar, which are not described herein again.

Fig. 7 is a schematic structural diagram of an audio noise reduction processing apparatus 700 according to the present application. As shown in fig. 7, the audio noise reduction processing apparatus 700 may include: at least one processor 701, a memory 702. The device may be a device (e.g., an earphone) having an active noise reduction function, or may be a chip, a chip module, or the like of the device having the active noise reduction function.

A memory 702 for storing programs. In particular, the program may include program code including computer operating instructions.

The memory 702 may comprise high-speed RAM memory, and may also include non-volatile memory (non-volatile memory), such as at least one disk memory.

The processor 701 is configured to execute computer-executable instructions stored in the memory 702 to implement the audio noise reduction processing method described in the foregoing method embodiments. The processor 701 may be a Central Processing Unit (CPU), an Application Specific Integrated Circuit (ASIC), or one or more Integrated circuits configured to implement the embodiments of the present Application.

The audio noise reduction processing apparatus 700 may further include a communication interface 703, so that the external device may perform communication interaction with the external device through the communication interface 703, where the external device may be, for example, the aforementioned device connected to an earphone, such as a mobile phone, a tablet, and the like. In a specific implementation, if the communication interface 703, the memory 702 and the processor 701 are implemented independently, the communication interface 703, the memory 702 and the processor 701 may be connected to each other through a bus and perform communication with each other. The bus may be an Industry Standard Architecture (ISA) bus, a Peripheral Component Interconnect (PCI) bus, an Extended ISA (EISA) bus, or the like. Buses may be classified as address buses, data buses, control buses, etc., but do not represent only one bus or type of bus.

Optionally, in a specific implementation, if the communication interface 703, the memory 702, and the processor 701 are integrated into a chip, the communication interface 703, the memory 702, and the processor 701 may complete communication through an internal interface.

The present application also provides a computer-readable storage medium, which may include: various media capable of storing program codes, such as a usb disk, a removable hard disk, a Read-Only Memory (ROM), a Random Access Memory (RAM), a magnetic disk or an optical disk, and in particular, the computer-readable storage medium stores program instructions, and the program instructions are used in the method in the foregoing embodiments.

The present application also provides a computer program product comprising executable instructions stored in a readable storage medium. The at least one processor of the audio noise reduction processing apparatus 700 may read the executable instructions from the readable storage medium, and the at least one processor executes the executable instructions to enable the audio noise reduction processing apparatus 700 to implement the audio noise reduction processing method provided by the various embodiments described above.

The application also provides a chip, wherein a computer program is stored on the chip, and when the computer program is executed by the chip, the audio noise reduction processing method provided by various embodiments is realized.

Other embodiments of the present application will be apparent to those skilled in the art from consideration of the specification and practice of the invention disclosed herein. This application is intended to cover any variations, uses, or adaptations of the invention following, in general, the principles of the application and including such departures from the present disclosure as come within known or customary practice within the art to which the invention pertains. It is intended that the specification and examples be considered as exemplary only, with a true scope and spirit of the application being indicated by the following claims.

It will be understood that the present application is not limited to the precise arrangements described above and shown in the drawings and that various modifications and changes may be made without departing from the scope thereof. The scope of the application is limited only by the appended claims.

Claims

1. An audio noise reduction processing method, the method comprising:

when the first audio signal is played by using the loudspeaker, a second audio signal collected by the error microphone is obtained; the second audio signal comprises: a first audio signal picked up through a target secondary channel between the loudspeaker and the error microphone;

compensating the third audio signal by using a compensation filter of an audio echo filter to obtain a fourth audio signal, wherein the compensation filter is used for compensating the audio echo filter so as to enable a target parameter of an error signal of the fourth audio signal and the stoped first audio signal to be less than or equal to a preset threshold value;

2. The method of claim 1, wherein the second audio signal further comprises: a noise signal of the internal environment, the method further comprising:

3. The method of claim 2, further comprising:

4. The method of claim 2, further comprising:

5. The method according to any one of claims 1-4, further comprising:

in the process of playing x (n) by using a loudspeaker, carrying out self-adaptive adjustment on the compensation filter according to an error signal between the audio signal compensated by the compensation filter and the audio signal collected by the error microphone to obtain a compensation filter coefficient;

configuring the compensation filter with the compensation filter coefficients.

6. The method of claim 5, wherein the target parameter is a mean square value, and the adaptively adjusting the compensation filter according to an error signal between the audio signal compensated by the compensation filter and the audio signal collected by the error microphone during the playing of x (n) by the speaker to obtain the compensation filter coefficient comprises:

processing the x (n) by adopting an audio echo filter to obtain an audio signal y (n); the audio echo filter is consistent with the frequency response of the simulated secondary channel;

acquiring an error signal e (n) between the z (n) and the m (n);

and utilizing the e (n) to carry out self-adaptive adjustment on the coefficient of the compensation filter until the mean square value of the error signal e (n) is less than or equal to a preset mean square value threshold value, and obtaining the coefficient of the compensation filter.

7. An audio noise reduction processing apparatus, characterized in that the apparatus comprises:

the acquisition module is used for acquiring a second audio signal acquired by the error microphone when the first audio signal is played by the loudspeaker; the second audio signal comprises: a first audio signal that is picked up through a target secondary channel between the speaker and the error microphone;

the processing module is used for processing the first audio signal by adopting an audio echo filter to obtain a third audio signal; the coefficient of the audio echo filter is obtained by pre-design;

the compensation module is used for compensating the third audio signal by using a compensation filter of an audio echo filter to obtain a fourth audio signal, and the compensation filter is used for compensating the audio echo filter so as to enable a target parameter of an error signal of the fourth audio signal and the stoped first audio signal to be smaller than or equal to a preset threshold value;

8. An audio noise reduction processing apparatus, comprising: a processor, and a memory communicatively coupled to the processor;

the memory stores computer-executable instructions;

the processor executes computer-executable instructions stored by the memory to implement the method of any of claims 1-6.

9. A computer-readable storage medium having computer-executable instructions stored thereon, which when executed by a processor, are configured to implement the method of any one of claims 1 to 6.

10. A computer program product, comprising a computer program which, when executed by a processor, implements the method of any one of claims 1 to 6.