CN112637741B

CN112637741B - Audio signal processing method, device, chip, electronic equipment and storage medium

Info

Publication number: CN112637741B
Application number: CN202011602177.0A
Authority: CN
Inventors: 刘佳泽; 王宇飞
Original assignee: Guangzhou Kugou Computer Technology Co Ltd
Current assignee: Guangzhou Kugou Computer Technology Co Ltd
Priority date: 2020-12-30
Filing date: 2020-12-30
Publication date: 2023-03-10
Anticipated expiration: 2040-12-30
Also published as: CN112637741A

Abstract

The disclosure provides a processing method, a processing device, a chip, electronic equipment and a storage medium of an audio signal, and belongs to the technical field of electronics. The method comprises the following steps: acquiring a bass signal; superposing the audio signal and the bass signal to obtain an audio superposed signal; and performing phase shift processing on the audio superposed signal to obtain a processed audio signal, wherein the phase of the high-tone signal in the processed audio signal is shifted backwards relative to the phase of the bass signal. The low-order filter is small in calculation amount and can be applied to a low-power-consumption platform. The bass signal extracted from the audio signal is superimposed on the original audio signal, so that the bass signal of the audio signal is enhanced. For the audio frequency superimposed signal with strengthened bass, the phase of the high pitch signal is shifted backwards relative to the phase of the low pitch signal by phase shifting processing, when a user listens to the audio frequency signal processed by the method, the low pitch signal can be obviously sensed, and the bass strengthening effect is better.

Description

Audio signal processing method, device, chip, electronic equipment and storage medium

Technical Field

The present disclosure relates to the field of electronic technologies, and in particular, to a method, an apparatus, a chip, an electronic device, and a storage medium for processing an audio signal.

Background

In modern life, song appreciation has become a common way of leisure and entertainment for users. In order to meet the user's requirements for the playback quality of a song, bass enhancement of the audio signal of the song is required. The earphone device is used as a common accessory when a user listens to songs, the size of the earphone device is smaller and smaller along with the trend of miniaturization of electronic equipment, the size of a battery is correspondingly reduced, and in order to guarantee the endurance time, the power consumption of the earphone device is often reduced. In a low power consumption scenario, how to process an audio signal to obtain a bass-enhanced sound effect becomes a concern for those skilled in the art.

In the field of audio processing, bass enhancement methods include IIR (Infinite Impulse Response) filter type enhancement, FIR (Finite Impulse Response) filter type enhancement, virtual bass enhancement, and the like. The IIR filter type enhancing method is low in calculation complexity and can run on a low-power-consumption platform, but the calculation result is not stable enough, the bass enhancing effect is not good, and even the method cannot be realized; the FIR filter type enhancement method needs to use a higher-order FIR filter, has higher computational complexity and is difficult to apply to a low-power platform; the virtual bass enhancement method needs to be matched with an IIR filter or an FIR filter, has high computational complexity and is difficult to apply to a low-power platform.

In order to achieve a bass enhancement effect on a low power platform, it is highly desirable to provide a method for processing an audio signal.

Disclosure of Invention

The embodiment of the disclosure provides an audio signal processing method, an audio signal processing device, an audio signal processing chip, an electronic device and a storage medium, which can achieve a bass enhancement effect on a low-power platform. The technical scheme is as follows:

in a first aspect, an audio signal processing chip is provided, the chip comprising: the device comprises a memory, a processor, an adder and a multiplier, wherein the memory, the processor, the adder and the multiplier are connected with a bus;

the memory is used for storing the audio signal, the first filtering parameter and the second filtering parameter;

the adder and the multiplier are used for forming a first filter and a second filter;

the processor is used for providing the first filtering parameter for the first filter, so that the first filter can extract a bass signal with the frequency smaller than a preset frequency from the audio signal, and superpose the bass signal and the audio signal to obtain an audio superposed signal;

the processor is further configured to provide the second filtering parameter to the second filter, so that the second filter can perform phase shift processing on the audio superposition signal to obtain a processed audio signal, the phase of a high-pitch signal in the processed audio signal is shifted backward relative to a low-pitch signal, and the high-pitch signal is a signal with a frequency greater than the preset frequency.

In another embodiment of the present disclosure, the chip further comprises: a parameter update interface connected to the bus;

the parameter updating interface is configured to receive an updated first filtering parameter and an updated second filtering parameter sent by an external device, and send the updated first filtering parameter and the updated first filtering parameter to the memory for storage, or,

the parameter updating interface is used for receiving audio attribute information sent by external equipment, sending the audio attribute information to the processor, calculating an updated first filtering parameter and an updated second filtering parameter by the processor according to the audio attribute information, and sending the updated first filtering parameter and the updated second filtering parameter to the memory for storage.

In another embodiment of the present disclosure, the first filtering parameter is a cut-off frequency of the first filter, and the first filtering parameter is smaller than a cut-off frequency of a bass signal to be enhanced in the audio signal.

In another embodiment of the present disclosure, the second filtering parameter is a phase shift parameter of the second filter, and the second filtering parameter is determined according to an order of the second filter and a preset maximum phase shift angle.

In another embodiment of the present disclosure, the determining process of the second filtering parameter is:

constructing a one-dimensional array, wherein the array comprises elements with the same number as the order number, the elements are represented by a complex expression taking a first numerical value as a variable, the first numerical value is determined according to an initial value of the first numerical value, an initial value of a second numerical value and a cyclic expression, the initial values of the first numerical value and the second numerical value are determined according to the order number and the maximum angle, and the cyclic expression is used for representing the relation between the first numerical values of different cycle times;

and carrying out inverse real discrete Fourier transform on each element in the array to obtain the second filtering parameter.

In another embodiment of the present disclosure, the first filter is a first order butterworth filter and the second filter is a finite impulse response FIR filter.

In a second aspect, an electronic device is provided, the electronic device comprising: the audio processing chip comprises a memory, a processor, an adder and a multiplier, wherein the memory, the processor, the adder and the multiplier are connected with a bus;

the processor is used for providing the first filtering parameter for the first filter, so that the first filter can extract a bass signal with a frequency smaller than a preset frequency from the audio signal, and superpose the bass signal and the audio signal to obtain an audio superposed signal;

the processor is further configured to provide the second filter parameter to the second filter, so that the second filter can perform phase shift processing on the audio superposition signal to obtain a processed audio signal, the phase of the high-pitch signal in the processed audio signal is shifted backward relative to the low-pitch signal, and the high-pitch signal is a signal with a frequency greater than the preset frequency.

In a third aspect, a method for processing an audio signal is provided, the method including:

processing the audio signal to obtain a bass signal with frequency less than a preset frequency;

carrying out superposition processing on the audio signal and the bass signal to obtain an audio superposed signal;

and performing phase shift processing on the audio superposed signal to obtain a processed audio signal, wherein the phase of a high-pitch signal in the processed audio signal is shifted backwards relative to the phase of a low-pitch signal, and the high-pitch signal is a signal with frequency greater than the preset frequency.

In another embodiment of the present disclosure, the superimposing the audio signal and the bass signal to obtain an audio superimposed signal includes:

and superposing the audio signal and the signal with the same playing time in the bass signal to obtain the audio superposed signal.

In another embodiment of the disclosure, the phase-shifting the audio superposition signal to obtain a processed audio signal includes:

determining phase shift angles corresponding to signals with different frequencies in the audio superposed signal;

and performing phase shift processing on the audio superposed signals according to phase shift angles corresponding to signals with different frequencies in the audio superposed signals to obtain the processed audio signals.

In another embodiment of the present disclosure, the determining phase shift angles corresponding to signals of different frequencies in the audio superposition signal includes:

acquiring the highest frequency and the lowest frequency of signals in the audio superposed signals;

and determining phase shift angles corresponding to signals with different frequencies according to the highest frequency, the lowest frequency and a preset phase shift angle.

In another embodiment of the present disclosure, the determining phase shift angles corresponding to signals of different frequencies according to the highest frequency, the lowest frequency and a preset phase shift angle includes:

constructing a first linear function according to the corresponding relation between the lowest frequency and the minimum phase shift angle and the corresponding relation between the highest frequency and the maximum phase shift angle, wherein the first linear function is used for indicating that the frequency and the phase shift angle form positive correlation;

and determining corresponding phase shift angles of the signals with different frequencies based on the first linear function.

In another embodiment of the present disclosure, the performing phase shift processing on the audio frequency superimposed signal according to phase shift angles corresponding to signals with different frequencies in the audio frequency superimposed signal to obtain the processed audio frequency signal includes:

and according to the phase shift angles corresponding to the signals with different frequencies in the audio superposed signal, the phase of the audio superposed signal is shifted backwards to obtain the processed audio signal.

constructing a second linear function according to the corresponding relation between the lowest frequency and the maximum phase shift angle and the corresponding relation between the highest frequency and the minimum phase shift angle, wherein the second linear function is used for indicating that the frequency and the phase shift angle form negative correlation;

and determining the corresponding phase shift angle of the signal with different frequencies based on the second linear function and different frequencies.

and according to the phase shift angles corresponding to the signals with different frequencies in the audio superposed signal, advancing the phase of the audio superposed signal to obtain the processed audio signal.

In a fourth aspect, an apparatus for processing an audio signal is provided, the apparatus comprising:

the first processing module is used for processing the audio signal to obtain a bass signal with frequency less than the preset frequency;

the second processing module is used for carrying out superposition processing on the audio signal and the bass signal to obtain an audio superposition signal;

and the third processing module is used for performing phase shift processing on the audio superposed signal to obtain a processed audio signal, wherein the phase of a high-pitch signal in the processed audio signal is shifted backwards relative to the phase of a low-pitch signal, and the high-pitch signal is a signal with the frequency greater than the preset frequency.

In another embodiment of the present disclosure, the second processing module is configured to superimpose signals with the same playing time in the audio signal and the bass signal, so as to obtain the audio superimposed signal.

In another embodiment of the present disclosure, the third processing module is configured to determine phase shift angles corresponding to signals of different frequencies in the audio superposition signal; and performing phase shift processing on the audio superposed signal according to phase shift angles corresponding to signals with different frequencies in the audio superposed signal to obtain the processed audio signal.

In another embodiment of the present disclosure, the third processing module is configured to obtain the highest frequency and the lowest frequency of the signal in the audio superposition signal; and determining phase shift angles corresponding to signals with different frequencies according to the highest frequency, the lowest frequency and a preset phase shift angle.

In another embodiment of the present disclosure, the third processing module is configured to construct a first linear function according to the corresponding relationship between the lowest frequency and the minimum phase shift angle and the corresponding relationship between the highest frequency and the maximum phase shift angle, where the first linear function is used to indicate that the frequency and the phase shift angle are in positive correlation; and determining phase shift angles corresponding to signals of different frequencies based on the first linear function.

In another embodiment of the disclosure, the third processing module is configured to, according to a phase shift angle corresponding to a signal with a different frequency in the audio superimposed signal, shift a phase of the audio superimposed signal backward to obtain the processed audio signal.

In another embodiment of the present disclosure, the third processing module is configured to construct a second linear function according to the corresponding relationship between the lowest frequency and the maximum phase shift angle and the corresponding relationship between the highest frequency and the minimum phase shift angle, where the second linear function is used to indicate that the frequency and the phase shift angle are in negative correlation; and determining the corresponding phase shift angle of the signal with different frequencies based on the second linear function and different frequencies.

In another embodiment of the disclosure, the third processing module is configured to advance a phase of the audio superposition signal according to a phase shift angle corresponding to a signal with a different frequency in the audio superposition signal, so as to obtain the processed audio signal.

In a fifth aspect, a computer-readable storage medium is provided, in which at least one program code is stored, and the at least one program code is loaded and executed by a processor to implement the method for processing an audio signal according to the third aspect.

The technical scheme provided by the embodiment of the disclosure has the following beneficial effects:

and the low-order filter is small in calculation amount and can be applied to a low-power-consumption platform. The bass signal extracted from the audio signal is superimposed on the original audio signal so that the bass signal of the audio signal is enhanced. For the audio frequency superimposed signal with strengthened bass, the phase of the high pitch signal is shifted backwards relative to the phase of the low pitch signal by phase shifting processing, when a user listens to the audio frequency signal processed by the method, the low pitch signal can be obviously sensed, and the bass strengthening effect is better.

Drawings

In order to more clearly illustrate the technical solutions in the embodiments of the present disclosure, the drawings needed to be used in the description of the embodiments are briefly introduced below, and it is obvious that the drawings in the following description are only some embodiments of the present disclosure, and it is obvious for those skilled in the art to obtain other drawings based on the drawings without creative efforts.

Fig. 1 is a schematic structural diagram of an audio signal processing chip according to an embodiment of the present disclosure;

fig. 2 is a schematic structural diagram of another audio signal processing chip provided in the embodiment of the present disclosure;

fig. 3 is a schematic structural diagram of another audio signal processing chip provided in the embodiment of the present disclosure;

fig. 4 is a schematic structural diagram of another audio signal processing chip provided in the embodiment of the present disclosure;

fig. 5 is a schematic structural diagram of an audio signal processing chip according to an embodiment of the present disclosure;

fig. 6 is a flowchart of a method for processing an audio signal according to an embodiment of the present disclosure;

fig. 7 is a flowchart of another audio signal processing method provided by an embodiment of the disclosure;

fig. 8 is a schematic structural diagram of an audio signal processing apparatus according to an embodiment of the present disclosure.

Detailed Description

To make the objects, technical solutions and advantages of the present disclosure more apparent, embodiments of the present disclosure will be described in detail with reference to the accompanying drawings.

It is to be understood that the terms "each," "a plurality," and "any" and the like, as used in the embodiments of the present disclosure, are intended to encompass two or more, each referring to each of the corresponding plurality, and any referring to any one of the corresponding plurality. For example, the plurality of words includes 10 words, and each word refers to each of the 10 words, and any word refers to any one of the 10 words.

An embodiment of the present disclosure provides an audio signal processing chip, referring to fig. 1, the chip includes: memory 101, processor 102, adder 103, and multiplier 104. The memory 101, the processor 102, the adder 103, and the multiplier 104 are connected to a bus, so that the memory 101, the processor 102, the adder 103, and the multiplier 104 can communicate with each other via the bus. The bus is generally a common communication trunk that transfers information between the various functional elements of the computer.

The adder 103 is a digital circuit and can perform digital addition. Multiplier 104 is an electronic device that performs the multiplication of two analog signals that are uncorrelated. The adder 103 and the multiplier 104 are used to form the first filter and the second filter in different logical composition ways. The first filter is a bass signal extraction filter for extracting a low frequency signal from an audio signal. The order of the first filter may be first order, second order, third order, etc., and it is preferable that the order of the first filter is first order in the embodiment of the present disclosure. The second filter is a phase-shifting filter for phase-shifting the audio signal. The order of the second filter may be eight to sixty-four, and preferably the order of the second filter is sixteen in the embodiment of the present disclosure.

The memory 101 is used not only to store various signaling, program codes, and the like, but also to store an input or chip-owned audio signal, a first filter parameter, a second filter parameter, and the like.

The first filter parameter is a parameter of a first filter formed by the adder and the multiplier in the embodiment of the present disclosure, and the first filter parameter may be a cut-off frequency of the first filter, and the like. The cutoff frequency is an index representing the frequency characteristic of the filter, and a frequency at which the amplitude of the output signal is reduced to 0.707 or 0.5 times the maximum value is referred to as the cutoff frequency, while the frequency of the input signal is changed while the amplitude of the input signal is kept unchanged. In the embodiment of the present disclosure, the first filtering parameter may be determined according to a cut-off frequency of a bass signal that needs to be enhanced in the audio signal to be processed, and is generally smaller than the cut-off frequency of the bass signal that needs to be enhanced in the audio signal. For example, the sampling rate of the audio signal is 48000Hz (hertz), the cut-off frequency of the bass signal to be enhanced is 120Hz, and the first filtering parameter may be chosen to be 100Hz.

The second filtering parameter is a phase shift parameter of a second filter formed by an adder and a multiplier in the implementation of the present disclosure, and the phase shift parameter is a parameter used for determining phase shift angles of audio signals of different frequencies during phase shift processing. The second filter parameter is a real number array having a length equal to an order of the second filter, for example, if the order of the second filter is sixteen, the second filter parameter is a real number array having a length of sixteen. The second filtering parameter may be determined according to the order of the second filter and a preset maximum phase shift angle, and the determining process of the second filtering parameter is as follows:

the method comprises the steps of firstly, constructing a one-dimensional array, wherein the array comprises elements with the same number as the order number, the elements are expressed by a complex expression with a first value as a variable, the first value is determined according to an initial value of the first value, an initial value of a second value and a loop expression, the initial values of the first value and the second value are determined according to the order number and a maximum angle, and the loop expression is used for expressing the relation between the first values of different loop times.

And secondly, performing inverse real discrete Fourier transform on each element in the array to obtain a second filtering parameter.

For the determination process of the second filtering parameter, the following will be explained in detail:

in the first step, a one-dimensional complex number array Z is defined, the length of the complex number array is the order N of the second filter, and the numerical type of the complex number array is not limited in the embodiments of the present disclosure.

In the second step, two variables K (corresponding to the second value) and H (corresponding to the first value) are defined, and the numerical types of the variables K and H are not limited in the embodiments of the present disclosure. Setting K to an initial value of

Wherein pi is the circumference ratio. The initial value of H is set to 2 π.

And thirdly, defining the values of loop variables i, i are 1 to N, namely, performing a first loop when i =1, performing a second loop when i =2, and performing an Nth loop when i = N.

The fourth step, define H _i+1 ＝H _i -K _i ,

And step five, defining Z (i) = COS (H) + SIN (H) j, wherein COS is a cosine calculation symbol, SIN is a sine calculation symbol, the unit of H is radian, and j is the unit of an imaginary part in the complex number. This equation is also a complex expression, i.e., Z = a + Bj, where Z is a complex number, a is the real part, and B is the imaginary part.

And step six, circulating i from 1 to N to obtain Z (1), Z (2) \8230;, Z (N).

And seventhly = IRDFT (Z) is calculated, wherein IRDFT is inverse real discrete Fourier transform, and Y is the output result of the transform. According to the principle of inverse real discrete fourier transform, M complex numbers are input, and M real numbers are output, so that Y is a real number array with a length of N, and Y is a second filtering parameter.

Floating point calculations may need to be performed during the design phase of the first filter and the second filter, and IRDFT calculations are needed, which is a process that uses more memory and consumes more computational resources of the processor. However, once the sound sampling rate, the bass cut-off frequency to be enhanced, and the order of the second filter are determined, the coefficients of the first filter and the second filter will remain unchanged, and therefore, the embodiments of the disclosure may determine the first filter coefficient and the second filter coefficient in advance on a high-performance computing platform, such as a smartphone, a personal computer, and the like, and then solidify the first filter coefficient and the second filter coefficient into the memory of a low-power platform after the calculation, and only need to obtain the first filter coefficient and the second filter coefficient from the memory during normal use, so as to implement bass enhancement.

The processor 102 is configured to provide the first filtering parameter to the first filter, so that the first filter can process the audio signal to obtain a bass signal, and superimpose the bass signal and the audio signal to obtain an audio superimposed signal.

The processor 102 is further configured to provide the second filter parameter to the second filter, so that the second filter can perform a phase shift process on the audio superposition signal to obtain a processed audio signal, where a phase of the high-tone signal in the processed audio signal is shifted backward with respect to a phase of the bass signal. In the field of electronic technology, phase-shifting refers to adjusting the acquisition time of an audio signal with reference to the original acquisition time of the audio signal, so that the adjusted acquisition time of the audio signal is later than the original acquisition time.

In another embodiment of the present disclosure, referring to fig. 2, the chip further includes a parameter updating interface 105, which is connected to the bus, and is capable of updating the first filtering parameter and the second filtering parameter, so that the chip can continuously meet the bass enhancement requirements of users for different audio signals. The parameter updating interface updates the first filtering parameter and the second filtering parameter in two ways, including but not limited to:

the first mode is that the parameter updating interface receives updated first filtering parameters and updated second filtering parameters sent by the external device, and sends the updated first filtering parameters and the updated first filtering parameters to the memory for storage.

And the processor calculates an updated first filtering parameter and an updated second filtering parameter according to the audio attribute information, and sends the updated first filtering parameter and the updated second filtering parameter to the memory for storage.

In another embodiment of the present disclosure, referring to fig. 3, the chip further comprises an audio input interface 106 and an audio output interface 107, the audio input interface 106 and the audio output interface 107 being connected to the bus. The audio input interface 106 is used for inputting audio signals to the chip; the audio output interface 107 is used for outputting the processed audio signal.

In another embodiment of the present disclosure, referring to fig. 4, when the audio input interface 106 and the audio output interface 107 are analog signal interfaces, the chip further includes: a digital-to-analog converter ADC108 unit and an analog-to-digital converter DAC unit 109, the ADC unit 108 and DAC unit 109 being connected to the bus. The ADC unit 108 is configured to convert the audio signal from an analog signal to a digital signal, and the DAC unit 109 is configured to convert the processed audio signal from a digital signal to an analog signal.

In another embodiment of the present disclosure, in order to improve data access efficiency, the memory 101 includes a first storage unit and a second storage unit (not shown in the drawings). The first storage unit is used for storing audio signals and the like, and the first storage unit can be a Random Access Memory (RAM) and the like; the second storage unit is used for storing the first filtering parameter, the second filtering parameter and the like.

In another embodiment of the present disclosure, referring to fig. 5, the chip further includes: a clock module 110, the clock module 110 being connected to the bus. The clock module 110 is configured to provide clock signals for the functional modules of the chip, where the clock signals are used for clock synchronization of the functional modules of the chip.

In another embodiment of the disclosure, the first filter is a first order butterworth filter and the second filter is a FIR filter.

It should be noted that the butterworth filter is one of IIR filters. For an IIR filter, different types (e.g., butterworth, elliptic, etc.), filter types (e.g., high pass, low pass, etc.), different orders, etc. do not generally change the processing structure of the IIR filter on the signal, and only the coefficients of the IIR filter change. The butterworth filter described above is a standard IIR filter that employs the coefficient design method of the embodiments of the present disclosure.

It should be noted that, the FIR filter in the embodiment of the present disclosure also adopts a standard FIR filter structure, and only proposes a novel and special FIR coefficient design method, which is used in combination with a specific IIR to boost bass. Therefore, the hardware structure of the FIR filter is not improved, and only the coefficient design method of the embodiment of the present disclosure is adopted.

When only the Butterworth filter is used, the phase delay of the Butterworth filter does not have a good effect on bass enhancement, and after the FIR filter is introduced, the phase of the high-pitch signal is greatly shifted backwards, so that the overall bass enhancement effect in the embodiment of the disclosure can be highlighted.

The embodiment of the present disclosure provides an electronic device, which may be a bluetooth headset, an intelligent speaker, an MP3 (Moving Picture Experts Group Audio Layer III, standard Audio Layer 3 for dynamic image expert compression), an MP4 (Moving Picture Experts Group Audio Layer IV, standard Audio Layer 4 for dynamic image expert compression), and other low power devices, and the embodiment of the present disclosure does not specifically limit the product type of the electronic device. The electronic device includes: the casing and audio frequency processing chip. Referring to fig. 1, the audio processing chip includes a memory, a processor, an adder, and a multiplier, and the memory, the processor, the adder, and the multiplier are connected to a bus.

the processor is used for providing the first filtering parameter for the first filter, so that the first filter can extract a bass signal with frequency less than preset frequency from the audio signal, and superpose the bass signal and the audio signal to obtain an audio superposed signal;

the processor is further configured to provide the second filtering parameter to the second filter, so that the second filter can perform phase shift processing on the audio superposition signal to obtain a processed audio signal, a phase of a high-pitch signal in the processed audio signal is shifted backward relative to a phase of a low-pitch signal, and the high-pitch signal is a signal with a frequency greater than a preset frequency.

In another embodiment of the disclosure, the first filtering parameter is a cut-off frequency of the first filter, the first filtering parameter being smaller than a cut-off frequency of a bass signal to be enhanced in the audio signal.

In addition, the audio signal processing chip in the electronic device according to the embodiment of the present disclosure further includes other functional modules shown in fig. 2 to 5, which refer to the audio signal processing chip shown in fig. 2 to 5, and details are not repeated here.

The embodiment of the present disclosure provides a method for processing an audio signal, referring to fig. 6, the method provided by the embodiment of the present disclosure includes:

601. and processing the audio signal to obtain a bass signal with the frequency less than the preset frequency.

The preset frequency is a cut-off frequency indicated by the first filtering parameter, and the preset frequency may be 100Hz, 120Hz, or the like.

602. And carrying out superposition processing on the audio signal and the bass signal to obtain an audio superposed signal.

603. And performing phase shift processing on the audio superposed signal to obtain a processed audio signal.

The phase of the high-pitch signal in the processed audio signal is shifted backward relative to the phase of the low-pitch signal, and the high-pitch signal is a signal with frequency greater than preset frequency.

The method provided by the embodiment of the disclosure adopts the low-order filter, has small calculation amount and can be applied to a low-power-consumption platform. The bass signal extracted from the audio signal is superimposed on the original audio signal so that the bass signal of the audio signal is enhanced. For the audio frequency superimposed signal with strengthened bass, the phase of the high pitch signal is shifted backwards relative to the phase of the low pitch signal by phase shifting processing, when a user listens to the audio frequency signal processed by the method, the low pitch signal can be obviously sensed, and the bass strengthening effect is better.

All the above optional technical solutions may be combined arbitrarily to form optional embodiments of the present disclosure, and are not described in detail herein.

The embodiment of the present disclosure provides a method for processing an audio signal, referring to fig. 7, taking an electronic device as an example to execute the embodiment of the present disclosure, a flow of the method provided by the embodiment of the present disclosure includes:

701. when an audio signal is input, the electronic device copies the audio signal.

Since the audio signal processing method provided by the embodiment of the present disclosure needs to overlap a bass signal extracted from an audio signal with the audio signal, in order to ensure that an unprocessed original audio signal can be acquired, when the audio signal is input through the audio input interface, the electronic device needs to copy the audio signal to obtain two identical audio signals, where one of the audio signals is used to extract the bass signal therefrom, and the other is used to overlap the extracted bass signal. For the audio signal to be superimposed, the electronic device may store it in memory for later use.

702. The electronic equipment processes the audio signal to obtain a bass signal with frequency less than the preset frequency.

The electronic device processes the audio signal with a first filter. By processing the audio signal, the electronic device extracts a signal with a frequency less than a preset frequency from the audio signal to obtain a bass signal. The duration of the bass signal is the same as the duration of the audio signal.

703. The electronic equipment carries out superposition processing on the audio signal and the bass signal to obtain an audio superposition signal.

Because the audio signal and the bass signal both have the playing time, the electronic device can superimpose the audio signal and the signal with the same playing time in the bass signal to obtain an audio superimposed signal. Since the bass signal is derived from the audio signal and the frequencies of the audio signal and the bass signal with the same playing time are the same, the audio signal and the bass signal with the same playing time are superimposed, and actually, the amplitudes of the audio signal and the bass signal with the same playing time are superimposed. By performing the superposition processing on the audio signal and the bass signal, the bass signal of the audio superposition signal can be enhanced.

704. The electronic equipment performs phase shift processing on the audio superposed signal to obtain a processed audio signal.

When the electronic device performs phase shift processing on the audio frequency superposed signal to obtain a processed audio frequency signal, the following method can be adopted:

7041. the electronic device determines phase shift angles corresponding to signals of different frequencies in the audio superimposed signal.

When the electronic equipment determines the phase shift angles corresponding to the signals with different frequencies in the audio superposed signal, the highest frequency and the lowest frequency of the signals in the audio superposed signal can be obtained, and then the phase shift angles corresponding to the signals with different frequencies are determined according to the highest frequency, the lowest frequency and the preset phase shift angle. The preset phase shift angle is a preset maximum offset angle of the audio signal, and the preset phase shift angle may be 180 degrees, 360 degrees, and the like.

When the electronic device determines the phase shift angles corresponding to the signals with different frequencies according to the highest frequency, the lowest frequency and the preset phase shift angle, the method includes, but is not limited to, the following two modes:

in the first mode, the electronic device constructs a first linear function according to the corresponding relationship between the lowest frequency and the minimum phase shift angle and the corresponding relationship between the highest frequency and the maximum phase shift angle, and then determines the phase shift angles corresponding to signals with different frequencies based on the first linear function. The first linear function is used to represent a positive correlation of frequency and phase shift angle.

Setting the lowest frequency to f ₁ The highest frequency is f ₂ The minimum phase shift angle is theta ₁ Maximum phase shift angle of theta ₂ And setting a first linear function y1= a1x + b1, where y1 denotes the frequency of the audio signal, x denotes the phase shift angle, and a1, b1 are coefficients to be determined. According to the corresponding relation between the lowest frequency and the minimum phase shift angle and the corresponding relation between the highest frequency and the maximum phase shift angle, determining that a1 is

b1 is

The first linear function thus constructed

In the second mode, the electronic device constructs a second linear function according to the corresponding relationship between the lowest frequency and the maximum phase shift angle and the corresponding relationship between the highest frequency and the minimum phase shift angle, and then determines the phase shift angles corresponding to signals with different frequencies based on the second linear function and different frequencies. The second linear function is used to indicate that the frequency and the phase shift angle are inversely related.

Setting the lowest frequency to f ₁ Maximum frequency of f ₂ The minimum phase shift angle is theta ₁ Maximum phase shift angle of theta ₂ And setting a second linear function y2= a2x + b2, where y2 denotes the frequency of the audio signal, x denotes the phase shift angle, and a2, b2 are the coefficients to be determined. According to the corresponding relation between the lowest frequency and the maximum phase shift angle and the corresponding relation between the highest frequency and the minimum phase shift angle, determining that a2 is

b2 is

The second linear function thus constructed

7042. And the electronic equipment performs phase shift processing on the audio superposed signal according to the corresponding phase shift angles of the signals with different frequencies in the audio superposed signal to obtain a processed audio signal.

And according to the corresponding phase shift angles of the signals with different frequencies in the audio superposed signals, the electronic equipment adopts a second filter to perform phase shift processing on the signals with different frequencies in the audio superposed signals to obtain processed audio signals. Due to the adoption of the two modes, the corresponding phase shift angles of the signals with different frequencies in the audio superposed signal are different, the larger the frequency of the first mode is, the larger the phase shift angle is, and in order to realize the bass enhancement effect, the phase of the signals with different frequencies in the audio superposed signal needs to be shifted backwards; the larger the frequency of the second method is, the smaller the phase shift angle is, and in order to realize the bass enhancement effect, it is necessary to advance the phases of the signals of different frequencies in the audio superimposed signal.

Therefore, in the first case, the electronic device will perform phase-shifting backward on the audio superimposed signal according to the phase-shifting angles corresponding to the signals with different frequencies in the audio superimposed signal, so as to obtain the processed audio signal. For example, the audio signal is shifted from the lowest frequency (0 Hz) to the highest frequency by a predetermined phase shift angle of 0-360 degrees, i.e., the 0Hz signal is phase-shifted backward by 0 degrees (unchanged), \8230; the signal phase of the highest frequency is phase-shifted backward by 360 degrees (one cycle), while the amplitude (response) of each frequency remains unchanged.

For the second case, the electronic device shifts the phase of the audio superimposed signal forward according to the phase shift angle corresponding to the signal with different frequency in the audio superimposed signal, so as to obtain the processed audio signal.

After the audio signal is processed through the step, the phase of the high-frequency signal in the processed audio signal is shifted backwards relative to the phase of the low-frequency signal, and a user can obviously perceive the low-frequency signal in psychoacoustics, so that the effect of bass enhancement is achieved.

705. The electronic device outputs the processed audio signal.

And after the audio signal is processed, the electronic equipment outputs the processed audio signal through the audio output interface.

The method provided by the embodiment of the disclosure can be used for bass enhancement on an ultra-low power consumption platform, and can achieve the effect of obviously enhancing bass on the basis of hardly reducing the endurance time of the low power consumption platform by combining a low-order low-precision IIR filter and a low-order phase-shifting FIR filter. Certainly, except that the bass enhancement can be carried out on the ultra-low power consumption platform, the system can also be used on other platforms, when the system is used on other platforms, the bass enhancement effect is ensured, meanwhile, the power consumption on other platforms is greatly reduced, and the performance of other platforms is improved.

For the whole process of the above audio signal processing, the following process can be referred to:

1. a1 st order butterworth low pass filter is designed, with a cut-off frequency of 100Hz, named BUTT. Because the coefficient of the Butterworth low-pass IIR filter of the 1 st order is less, the stability and the precision of the filter can be ensured under the condition of fixed-point calculation.

2. A 16 th order FIR filter is designed, named PHASFT.

3. The input audio signal is named as IN and temporarily stored, a part of the input audio signal is copied and processed by BUTT, a bass signal LP is obtained, the extracted bass signal LP is accumulated back to the original input signal IN, and a bass enhanced signal TMP is obtained.

4. The TMP signal is PHASFT processed to obtain the final signal OUT. After the PHASFT processing, the phase of the high pitch signal is shifted backward relative to the phase of the low pitch signal, so that the user can obtain the bass with more obvious perception in psychoacoustics, and the final bass enhancement effect is better.

5. And outputting the OUT signal.

According to the method provided by the embodiment of the disclosure, the bass signal extracted from the audio signal is superimposed to the original audio signal, so that the bass signal of the audio signal is strengthened and enhanced. For the audio frequency superposed signal with strengthened bass, the phase of the high-pitch signal is backward shifted relative to the phase of the low-pitch signal by phase shifting processing, when a user listens to the audio frequency signal processed by the method, the low-pitch signal can be obviously sensed, and the bass strengthening effect is better.

Referring to fig. 8, an embodiment of the present disclosure provides an apparatus for processing an audio signal, where the apparatus is disposed in an electronic device, and the apparatus includes:

the first processing module 801 is configured to process an audio signal to obtain a bass signal with a frequency less than a preset frequency;

the second processing module 802 is configured to perform superposition processing on the audio signal and the bass signal to obtain an audio superposed signal;

the third processing module 803 is configured to perform phase shift processing on the audio superimposed signal to obtain a processed audio signal, where a phase of a treble signal in the processed audio signal is shifted backward with respect to a phase of a bass signal, and the treble signal is a signal with a frequency greater than a preset frequency.

In another embodiment of the present disclosure, the second processing module 802 is configured to superimpose signals with the same playing time in the audio signal and the bass signal, so as to obtain an audio superimposed signal.

In another embodiment of the present disclosure, the third processing module 803 is configured to determine phase shift angles corresponding to signals of different frequencies in the audio superposition signal; and performing phase shift processing on the audio superposed signal according to the phase shift angles corresponding to the signals with different frequencies in the audio superposed signal to obtain a processed audio signal.

In another embodiment of the present disclosure, the third processing module 803 is configured to obtain the highest frequency and the lowest frequency of the signal in the audio superposition signal; and determining phase shift angles corresponding to signals with different frequencies according to the highest frequency, the lowest frequency and the preset phase shift angle.

In another embodiment of the present disclosure, the third processing module 803 is configured to construct a first linear function according to a corresponding relationship between a lowest frequency and a minimum phase shift angle and a corresponding relationship between a highest frequency and a maximum phase shift angle, where the first linear function is used to indicate that the frequency and the phase shift angle have a positive correlation; based on the first linear function, phase shift angles corresponding to signals of different frequencies are determined.

In another embodiment of the present disclosure, the third processing module 803 is configured to perform phase-shifting backward on the audio superimposed signal according to a phase-shifting angle corresponding to a signal with a different frequency in the audio superimposed signal, so as to obtain a processed audio signal.

In another embodiment of the present disclosure, the third processing module 803 is configured to construct a second linear function according to the corresponding relationship between the lowest frequency and the maximum phase shift angle and the corresponding relationship between the highest frequency and the minimum phase shift angle, where the second linear function is used to indicate that the frequency and the phase shift angle are in negative correlation; and determining the corresponding phase shift angle of the signal with different frequencies based on the second linear function and different frequencies.

In another embodiment of the present disclosure, the third processing module 803 is configured to advance the phase of the audio superposition signal according to the phase shift angle corresponding to the signal with different frequency in the audio superposition signal, so as to obtain a processed audio signal.

In summary, the apparatus provided in the embodiment of the present disclosure uses a low-order filter to calculate a smaller amount, and can be applied to a low-power platform. The bass signal extracted from the audio signal is superimposed on the original audio signal, so that the bass signal of the audio signal is enhanced. For the audio frequency superposed signal with strengthened bass, the phase of the high-pitch signal is backward shifted relative to the phase of the low-pitch signal by phase shifting processing, when a user listens to the audio frequency signal processed by the method, the low-pitch signal can be obviously sensed, and the bass strengthening effect is better.

With regard to the apparatus in the above-described embodiment, the specific manner in which each module performs the operation has been described in detail in the embodiment related to the method, and will not be elaborated here.

The disclosed embodiments provide a computer-readable storage medium having at least one program code stored therein, the at least one program code being loaded and executed by a processor to implement the method for processing an audio signal shown in fig. 6 or fig. 7.

The computer-readable storage medium provided by the embodiment of the disclosure adopts a low-order filter, has a small calculation amount, and can be applied to a low-power-consumption platform. The bass signal extracted from the audio signal is superimposed on the original audio signal, so that the bass signal of the audio signal is enhanced. For the audio frequency superimposed signal with strengthened bass, the phase of the high pitch signal is shifted backwards relative to the phase of the low pitch signal by phase shifting processing, when a user listens to the audio frequency signal processed by the method, the low pitch signal can be obviously sensed, and the bass strengthening effect is better.

It will be understood by those skilled in the art that all or part of the steps for implementing the above embodiments may be implemented by hardware, or may be implemented by a program instructing relevant hardware, where the program may be stored in a computer-readable storage medium, and the above-mentioned storage medium may be a read-only memory, a magnetic disk or an optical disk, etc.

The above description is intended to be exemplary only and not to limit the present disclosure, and any modification, equivalent replacement, or improvement made without departing from the spirit and scope of the present disclosure is to be considered as the same as the present disclosure.

Claims

1. An audio signal processing chip, characterized in that the chip comprises: the device comprises a memory, a processor, an adder and a multiplier, wherein the memory, the processor, the adder and the multiplier are connected with a bus;

the memory is used for storing the audio signal, the first filter parameter and the second filter parameter, and a first filter coefficient and a second filter coefficient are solidified in the memory;

the adder and the multiplier are used for forming a first filter and a second filter, the first filter is a first-order Butterworth filter, and the second filter is a finite impulse response FIR filter;

the processor is further configured to provide the second filter with the second filter parameters such that the second filter is capable of obtaining a highest frequency and a lowest frequency of a signal in the audio superposition signal; determining phase shift angles corresponding to signals with different frequencies according to the highest frequency, the lowest frequency and a preset phase shift angle; according to the phase shift angles corresponding to the signals with different frequencies in the audio superposed signal, the audio superposed signal is subjected to phase shift processing to obtain a processed audio signal, the phase of a high-pitch signal in the processed audio signal is shifted backwards relative to a low-pitch signal, and the high-pitch signal is a signal with a frequency greater than a preset frequency.

2. The chip of claim 1, wherein the chip further comprises: a parameter update interface connected to the bus;

3. The chip of claim 1, wherein the first filtering parameter is a cut-off frequency of the first filter, and wherein the first filtering parameter is smaller than a cut-off frequency of a bass signal to be enhanced in the audio signal.

4. The chip of claim 1, wherein the second filtering parameter is a phase shift parameter of the second filter, and the second filtering parameter is determined according to an order of the second filter and a preset maximum phase shift angle.

5. The chip of claim 4, wherein the second filter parameter is determined by:

constructing a one-dimensional array, wherein the array comprises elements with the same number as the order number, the elements are represented by a complex expression with a first value as a variable, the first value is determined according to an initial value of the first value, an initial value of a second value and a cyclic expression, the initial values of the first value and the second value are determined according to the order number and the maximum phase shift angle, and the cyclic expression is used for representing the relation between the first values of different cycle times;

6. An electronic device, characterized in that the electronic device comprises: the audio processing chip comprises a memory, a processor, an adder and a multiplier, wherein the memory, the processor, the adder and the multiplier are connected with a bus;

7. The electronic device of claim 6, wherein the first filtering parameter is a cut-off frequency of the first filter, and wherein the first filtering parameter is less than a cut-off frequency of a bass signal to be enhanced in the audio signal.

8. The electronic device of claim 6, wherein the second filtering parameter is a phase shift parameter of the second filter, and the second filtering parameter is determined according to an order of the second filter and a preset maximum phase shift angle.

9. A method of processing an audio signal, the method comprising:

determining phase shift angles corresponding to signals with different frequencies according to the highest frequency, the lowest frequency and a preset phase shift angle;

according to phase shift angles corresponding to signals with different frequencies in the audio superposed signal, performing phase shift processing on the audio superposed signal to obtain a processed audio signal, wherein the phase of a high-pitch signal in the processed audio signal is shifted backwards relative to the phase of a low-pitch signal, and the high-pitch signal is a signal with the frequency greater than the preset frequency;

the superposition processing is realized by a first filter, the phase shift processing is realized by a second filter, the first filter is a first-order Butterworth filter, the second filter is a finite impulse response FIR filter, and a first filter coefficient and a second filter coefficient are solidified in the memory.

10. The method of claim 9, wherein the superimposing the audio signal and the bass signal to obtain an audio superimposed signal comprises:

11. The method according to claim 9, wherein the determining phase shift angles corresponding to signals of different frequencies according to the highest frequency, the lowest frequency and a preset phase shift angle comprises:

12. The method of claim 11, wherein the performing phase shift processing on the audio frequency superposition signal according to the corresponding phase shift angles of the signals with different frequencies in the audio frequency superposition signal to obtain the processed audio signal comprises:

13. The method of claim 9, wherein determining phase shift angles corresponding to signals of different frequencies according to the highest frequency, the lowest frequency and a predetermined phase shift angle comprises:

14. The method of claim 13, wherein the phase-shifting the audio frequency superposition signal according to the phase-shifting angles corresponding to the signals with different frequencies in the audio frequency superposition signal to obtain the processed audio frequency signal comprises:

15. An apparatus for processing an audio signal, the apparatus comprising:

the second processing module is used for performing superposition processing on the audio signal and the bass signal to obtain an audio superposed signal;

the third processing module is used for acquiring the highest frequency and the lowest frequency of the signals in the audio superposed signals; determining phase shift angles corresponding to signals with different frequencies according to the highest frequency, the lowest frequency and a preset phase shift angle; according to phase shift angles corresponding to signals with different frequencies in the audio superposed signal, performing phase shift processing on the audio superposed signal to obtain a processed audio signal, wherein the phase of a high-pitch signal in the processed audio signal is shifted backwards relative to the phase of a low-pitch signal, and the high-pitch signal is a signal with the frequency greater than the preset frequency;

16. The apparatus of claim 15, wherein the second processing module is configured to superimpose signals with the same playing time in the audio signal and the bass signal to obtain the audio superimposed signal.

17. The apparatus of claim 15, wherein the third processing module is configured to construct a first linear function according to the corresponding relationship between the lowest frequency and the minimum phase shift angle and the corresponding relationship between the highest frequency and the maximum phase shift angle, and the first linear function is used to represent that the frequency and the phase shift angle are in positive correlation; and determining corresponding phase shift angles of the signals with different frequencies based on the first linear function.

18. The apparatus of claim 17, wherein the third processing module is configured to, according to a phase shift angle corresponding to a signal with a different frequency in the audio superimposed signal, shift a phase of the audio superimposed signal backward to obtain the processed audio signal.

19. The apparatus according to claim 15, wherein the third processing module is configured to construct a second linear function according to the corresponding relationship between the lowest frequency and the maximum phase shift angle and the corresponding relationship between the highest frequency and the minimum phase shift angle, the second linear function being used to indicate that the frequency and the phase shift angle are in negative correlation; and determining the corresponding phase shift angle of the signal with different frequencies based on the second linear function and different frequencies.

20. The apparatus of claim 19, wherein the third processing module is configured to advance a phase of the audio superposition signal according to a phase shift angle corresponding to a signal with a different frequency in the audio superposition signal, so as to obtain the processed audio signal.

21. A computer-readable storage medium, characterized in that at least one program code is stored in the storage medium, which is loaded and executed by a processor, to implement the method of processing an audio signal according to any one of claims 9 to 14.