CN118824261A - Audio processing method, chip and electronic device - Google Patents

Abstract

The application relates to the technical field of audio processing, and discloses an audio processing method, a chip and electronic equipment. The audio processing method of the present application includes: acquiring a first audio signal; generating a first harmonic signal based on the first audio signal; performing feature recognition on the first audio signal to determine the signal type of the first audio signal; determining the harmonic proportion of the first audio signal in real time according to the signal type of the first audio signal; and adding the original audio signal of the first audio signal and the first harmonic signal based on the harmonic proportion of the first audio signal to obtain a second audio signal. According to the scheme, the harmonic proportion can be regulated and controlled in real time according to the signal type of the first audio signal, so that a user can intuitively feel the low-frequency hearing of low-frequency audio of different types, and the playing effect of the audio signal is improved.

Description

Audio processing method, chip and electronic device

Technical Field

The present application relates to the field of audio processing technology, and in particular to an audio processing method, chip and electronic device.

Background Art

With the development of science and technology, portable electronic devices such as mobile phones and tablet computers have become commonly used electronic products in people's lives, and people use portable electronic devices to play audio signals in daily life.

Portable electronic devices are limited by their size and usually use micro speakers as sound-generating units. Micro speakers are limited by physical conditions and rules such as unit size and cabinet structure, and their cutoff frequencies are usually relatively high, generally 400-1000Hz. Therefore, the low-frequency signals below 400Hz reproduced usually have low clarity and severe distortion. However, the low-frequency signals in audio signals play an important role in the sound quality of audio signals. The fullness, thickness, and spatial sense of the sound are all represented by the low-frequency signals in the audio signals. For example: low-frequency signals of 100-150Hz affect the fullness of the timbre, low-frequency signals of 60-100Hz affect the thickness of the audio; and low-frequency signals of 20-60Hz affect the spatial sense of the timbre. Therefore, how to improve the ability of micro speakers to reproduce the low-frequency hearing of low-frequency signals is a problem that needs to be solved in the field of audio processing.

Summary of the invention

In order to improve the ability of a micro speaker to reproduce the low-frequency hearing sensation of low-frequency data, an embodiment of the present application provides an audio processing method, a chip, and an electronic device.

In a first aspect, an embodiment of the present application provides an audio processing method, the audio processing method comprising: acquiring a first audio signal; generating a first harmonic signal based on the first audio signal; performing feature recognition on the first audio signal to determine a signal type of the first audio signal; determining a harmonic ratio of the first audio signal in real time according to the signal type of the first audio signal; and based on the harmonic ratio of the first audio signal, summing an original audio signal of the first audio signal and the first harmonic signal to obtain a second audio signal.

The embodiment of the present application identifies the components in the first audio signal and adjusts the harmonic ratio in real time according to the signal type of the first audio signal so that the user can intuitively feel the low-frequency hearing sense of the low-frequency audio. The second audio signal with a low-frequency hearing sense not only retains the hearing sense corresponding to the high-frequency signal in the original audio signal of the first audio signal, but also can virtualize the bass hearing sense of the low-frequency signal in the first audio signal, thereby improving the playback effect of the audio signal.

In a possible implementation, the harmonic ratio of the first audio signal is determined in real time according to the signal type of the first audio signal, including: corresponding to the signal type of the first audio signal being a background sound signal, reducing the harmonic ratio of the first audio signal from a preset ratio to a first ratio; corresponding to the signal type of the first audio signal being a voice signal, reducing the harmonic ratio of the first audio signal from a preset ratio to a second ratio, wherein the first ratio is smaller than the second ratio.

In an embodiment of the present application, the harmonic ratio is adjusted in real time so that the harmonic signal component corresponding to the human voice signal in the low-frequency signal is smaller than the harmonic signal component corresponding to the background sound signal, thereby improving the clarity of the human voice signal in the low-frequency signal.

In a possible implementation, reducing the harmonic ratio of the first audio signal from a preset ratio to a first ratio includes: reducing the harmonic ratio of the first audio signal from a preset ratio to the first ratio based on a first speed. Reducing the harmonic ratio of the first audio signal from a preset ratio to a second ratio includes: reducing the harmonic ratio of the first audio signal from a preset ratio to the second ratio based on a second speed; wherein the first speed is less than the second speed.

In a possible implementation, performing feature recognition on the first audio signal to determine the signal type of the first audio signal includes: performing feature recognition on the first audio signal based on preset features to determine the signal type of the first audio signal, wherein the preset features include a zero crossing rate and a waveform factor.

In a possible implementation, obtaining a first audio signal includes: obtaining an original audio signal, filtering the original audio signal through a first filter group, and obtaining a first audio signal; generating a first harmonic signal based on the first audio signal includes: generating a second harmonic signal based on the first audio signal; and filtering the second harmonic signal through a second filter group to obtain a first harmonic signal.

In a possible implementation, the first filter group and the second filter group both include a high-pass filter and a low-pass filter, and both the high-pass filter and the low-pass filter are linear filters.

In one possible implementation, the linear filter comprises a Bessel filter.

In a possible implementation, the bandwidth of the first filter group is determined based on the audio bandwidth of the original audio signal and the bandwidth of a speaker of the electronic device; the bandwidth of the second filter group is determined based on the bandwidth of the speaker of the electronic device.

In one possible implementation, the bandwidth of the first filter group is determined based on the audio bandwidth of the original audio signal and the bandwidth of the speaker of the electronic device, including: the cutoff frequency of the high-pass filter in the first filter group is determined based on the lower limit frequency of the original audio signal and the low-frequency cutoff frequency of the speaker of the electronic device, and the cutoff frequency of the low-pass filter in the first filter group is determined based on the low-frequency cutoff frequency and the high-frequency cutoff frequency of the speaker of the electronic device.

In one possible implementation, the bandwidth of the second filter group is determined based on the bandwidth of the speaker of the electronic device, including: the cutoff frequency of the high-pass filter in the second filter group is determined based on the low-frequency cutoff frequency of the speaker of the electronic device, and the cutoff frequency of the low-pass filter in the second filter group is determined based on the high-frequency cutoff frequency of the speaker of the electronic device.

In a second aspect, an embodiment of the present application provides a chip, including a circuit, which is used to execute any audio processing method provided by the first aspect and various possible implementations of the first aspect.

In a third aspect, an embodiment of the present application provides an electronic device comprising the above-mentioned chip.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG1 is a schematic diagram showing a flow chart of an audio processing method according to an embodiment of the present application;

FIG2 is a schematic diagram showing a phase-frequency characteristic curve of a Bessel filter according to an embodiment of the present application;

FIG3 is a schematic diagram showing an amplitude-frequency characteristic curve of a Bessel filter according to an embodiment of the present application;

FIG4 is a schematic diagram showing a flow chart of an audio processing method according to an embodiment of the present application;

FIG5 shows a schematic structural diagram of an electronic device 1300 according to an embodiment of the present application.

DETAILED DESCRIPTION

The following first introduces the technical terms of this application.

Fundamental wave: refers to the sinusoidal wave component that is equal to the longest period of oscillation of periodic non-sinusoidal alternating current quantity. The frequency corresponding to this period is called the fundamental wave frequency.

Harmonic: refers to the components greater than the integer multiples of the fundamental frequency obtained by Fourier series decomposition of periodic non-sinusoidal alternating current. The frequency of the harmonic is equal to the integer multiples of the fundamental frequency. For example, a wave with 3 times the fundamental frequency is called the third harmonic, and a wave with 5 times the fundamental frequency is called the fifth harmonic.

Band-pass filter: A filter that can pass frequency components within a certain frequency range while attenuating frequency components in other frequency ranges to extremely low levels.

Cut-off frequency: used to describe the frequency characteristics of filters or amplifiers. When a filter or an amplifier keeps the amplitude of the input signal unchanged and changes the signal frequency to reduce the output signal to 0.707 times (corresponding to -3dB) or 0.5 times (corresponding to -6dB) of the maximum value, the frequency is called the cut-off frequency.

Harmonic ratio: the ratio of harmonic components to fundamental components.

Bessel filter: is a linear filter with linear phase response.

The illustrative embodiments of the present application include, but are not limited to, an audio processing method, a chip, and an electronic device.

It can be understood that the technical solution of the present application is applicable to electronic devices with speakers (such as micro speakers), for example, including but not limited to mobile phones, smart watches, televisions, tablet computers, wearable devices, vehicle-mounted devices, augmented reality (AR)/virtual reality (VR) devices, laptops, ultra-mobile personal computers (UMPC), netbooks, personal digital assistants (PDA), etc. The embodiments of the present invention do not impose any restrictions on the specific types of electronic devices.

It is understandable that micro speakers are limited by physical conditions and laws such as unit size and cabinet structure. The cutoff frequency is usually relatively high, generally 400Hz to 1000Hz. Therefore, low-frequency signals below 400Hz are usually not reproduced well, and the ability to restore low-frequency signals is relatively weak.

However, low-frequency signals easily resonate with the human body and are the most obvious audio signals perceived by the human body. For example, the fundamental frequencies of instruments such as drums and bass, as well as human voices, are generally below 400Hz. People can more clearly feel the impact of music and voices based on low-frequency signals. Therefore, the sound quality of the audio signal perceived by the user can be improved by improving the ability of micro speakers to reproduce the low-frequency hearing of low-frequency signals.

In some embodiments, in order to improve the ability of the micro-speaker to reproduce the low-frequency hearing sense of the low-frequency signal, a virtual bass method is used to compensate for the low-frequency hearing sense of the low-frequency signal, including: using a Butterworth filter to filter the original audio signal to obtain the fundamental signal of the original audio signal. Among them, the frequency of the harmonic signal can be set to be higher than 3 times the cutoff frequency of the micro-speaker. For example, the cutoff frequency of the micro-speaker is 400Hz, and the frequency of the fundamental signal is 100Hz, then the 4th harmonic of the fundamental signal is used as the harmonic signal of the virtual low-frequency signal bass effect, that is, the signal with four times the frequency of the fundamental signal is used as the harmonic signal of the virtual low-frequency signal bass effect. Finally, the virtual bass signal is added to the original audio signal to obtain an audio signal with a low-frequency hearing sense.

However, since the Butterworth filter is a nonlinear filter, the nonlinear filter changes the relative phase of each frequency component in the filtered signal, resulting in a phase deviation between the virtual bass signal and the original audio signal, such as the peak of the virtual bass signal does not correspond to the peak of the original audio signal. In addition, in the above scheme, the ratio of the addition of the virtual bass signal to the original audio signal is fixed, that is, regardless of whether the original audio signal is a human voice signal or a background sound signal, the ratio of the addition of the harmonic signal to the original audio signal is constant. When ensuring the clarity of the background sound signal, the human voice signal may be too thick or a little hoarse, and the clarity is low; when ensuring the clarity of the human voice signal, the low-frequency strength of the background sound signal may be insufficient.

In other embodiments, the original audio signal is divided into two paths, which are respectively passed through a high-pass filter and a low-pass filter. The high-frequency signal after the high-pass filter is directly retained after a preset time delay; the low-frequency signal after filtering by the low-pass filter needs to be processed by a harmonic generator and an energy control module to obtain a virtual bass signal. Finally, the virtual bass signal and the delayed high-frequency signal are superimposed to obtain an output signal. This virtual bass method also has the problem of a fixed ratio between the virtual bass signal and the original audio signal, that is, when ensuring the clarity of the background sound signal, the human voice signal may be too thick or a little hoarse, and the clarity is low. When ensuring the clarity of the human voice signal, the low-frequency strength of the background sound signal may be insufficient.

In order to solve the problem that the low-frequency signal reproduced by the above-mentioned micro-speaker has low clarity and severe distortion, an embodiment of the present application provides an audio processing method, which includes: obtaining a first audio signal; generating a first harmonic signal based on the first audio signal; performing feature recognition on the first audio signal to determine the signal type of the first audio signal; determining the harmonic ratio of the first audio signal in real time according to the signal type of the first audio signal; and based on the harmonic ratio of the first audio signal, summing the original audio signal of the first audio signal and the first harmonic signal to obtain a second audio signal.

The embodiment of the present application identifies the components in the first audio signal and adjusts the harmonic ratio in real time according to the signal type of the first audio signal so that the user can intuitively feel the low-frequency hearing of different types of low-frequency signals, so that the obtained second audio signal with low-frequency hearing not only retains the hearing corresponding to the high-frequency signal in the original audio signal of the first audio signal, but also can virtualize the bass hearing of the low-frequency signal in the first audio signal, thereby improving the playback effect of the audio signal.

In order to make the purpose, technical solutions and advantages of the present application clearer, the technical solutions in the embodiments of the present application are clearly and comprehensively described below with reference to the accompanying drawings, taking the video playback scenario as an example.

The following is a detailed description of the audio processing method provided in the embodiment of the present application. The audio processing method in the embodiment of the present application is applied to an electronic device having a micro speaker. FIG1 shows a schematic diagram of an audio processing method in the embodiment of the present application. The audio processing method includes:

101: Acquire a first audio signal.

In an embodiment of the present application, an original audio signal may be first obtained, and the original audio signal may be filtered through a first filter group to obtain a low-frequency signal for generating a harmonic signal, that is, a first audio signal for generating a harmonic signal. The original audio signal may be an original audio signal that needs to be played through a micro speaker, for example, the original audio signal may be music played by an audio playback application, the voice of a caller in a hands-free call, etc.

Wherein, the first filter group includes a high-pass filter and a low-pass filter, and the high-pass filter and the low-pass filter can be connected in series. In the case where the high-pass filter and the low-pass filter are connected in series, the frequency range of the signal that can be passed by the first filter is from the cutoff frequency of the high-pass filter to the cutoff frequency of the low-pass filter. In addition, since the high-pass filter and the low-pass filter are connected in series, the signal needs to pass through the high-pass filter first and then through the low-pass filter. This configuration will increase the delay of the signal and can offset the phase difference between the acquired first audio signal and the original audio signal.

It can be understood that the high-pass filter is used to perform high-pass filtering on the original audio signal to obtain a signal higher than the cutoff frequency of the high-pass filter; the low-pass filter is used to perform low-pass filtering on the original audio signal after high-pass filtering to obtain a signal lower than the cutoff frequency of the low-pass filter.

In the embodiment of the present application, the bandwidth of the first filter group may be determined according to the audio bandwidth of the original audio signal and the bandwidth of the micro speaker of the electronic device.

In some embodiments, the cutoff frequency of the low-pass filter in the first filter group can be determined according to the low-frequency cutoff frequency and the high-frequency cutoff frequency of the micro-speaker, and the cutoff frequency of the high-pass filter in the first filter group can be determined according to the lower limit frequency of the original audio signal and the low-frequency cutoff frequency of the micro-speaker. For example, the cutoff frequency of the low-pass filter can be greater than or equal to the low-frequency cutoff frequency of the micro-speaker and less than the high-frequency cutoff frequency of the micro-speaker, and the cutoff frequency of the high-pass filter can be greater than or equal to the lower limit frequency of the original audio signal and less than the low-frequency cutoff frequency of the micro-speaker.

For example, the frequency range of the original audio signal is 20Hz to 20kHz, the low-frequency cutoff frequency of the micro-speaker of the mobile phone is 400Hz, and the high-frequency cutoff frequency is 10KHz, that is, the micro-speaker of the mobile phone cannot reproduce the signal of 20Hz to 400Hz in the original audio signal well. In order to obtain the low-frequency signal of the original audio signal, the cutoff frequency of the high-pass filter in the first filter group can be set to be greater than or equal to the lower limit frequency of the original audio signal and less than the low-frequency cutoff frequency of the micro-speaker of the mobile phone, for example, 20Hz, and the cutoff frequency of the low-pass filter in the first filter group is set to be greater than or equal to the low-frequency cutoff frequency of the speaker of the electronic device and less than the high-frequency cutoff frequency of the speaker of the electronic device, such as 400Hz, that is, a bandpass of 20Hz to 400Hz is formed by the first filter group to filter the original audio signal to obtain a first audio signal for generating a virtual bass signal, wherein the frequency band range of the first audio signal satisfies 20Hz to 400Hz.

In the embodiment of the present application, the filter of the first filter group is a linear filter, such as a Bessel filter. It can be understood that the filter of the first filter group is a linear filter, and the audio signal filtered by the linear filter has a linear phase, which ensures that the relative phase relationship of each frequency component in the filtered first audio signal does not change, avoiding the problem of phase mismatch between the first audio signal and the original audio signal.

The transfer function of the low-pass filter in the first filter group is shown in formula (1):

Wherein, θ _n (0) and θ _n (s/ω ₀ ) are inverse Bessel polynomials, s is the frequency information, ω ₀ is the selected desired cutoff frequency, and s/ω ₀ is the phase information.

The phase-frequency characteristic curve of the Bessel low-pass filter is shown in Figure 2. The horizontal axis in Figure 2 is the ω of the Bessel low-pass filter, and the vertical axis is the phase of the Bessel low-pass filter. It can be seen from Figure 2 that the phase of the Bessel low-pass filter increases monotonically with the increase of ω, that is, the phase of the Bessel low-pass filter is a linear phase, that is, the phase of the signal filtered by the Bessel low-pass filter will not change.

The amplitude-frequency characteristic curve of the Bessel low-pass filter is shown in Figure 3. The horizontal axis in Figure 3 is the frequency of the Bessel low-pass filter, and the vertical axis is the amplitude of the signal allowed to pass by the Bessel low-pass filter. It can be seen from Figure 3 that the amplitude of the signal that can be passed by the Bessel low-pass filter is less than 0, and gradually decreases with the increase of frequency, that is, the Bessel low-pass filter can pass low-frequency signals and suppress high-frequency signals.

102: Generate a first harmonic signal based on the first audio signal.

In an embodiment of the present application, the first audio signal is output to a harmonic generator, so as to generate a second harmonic signal based on the first audio signal through the harmonic generator. Then, the second harmonic signal is filtered by a second filter group to obtain a first harmonic signal, i.e., a virtual bass signal. It can be understood that in the process of processing audio signals, high-frequency harmonics are usually related to the clarity of sound. The present application can enhance the brightness and clarity of the signal by adding the harmonic signal of the audio signal, thereby improving the auditory clarity and presence of the audio signal.

In some embodiments, if the frequency of the first audio signal is 20 Hz to 400 Hz, generally, in order to increase the clarity of the audio signal, one to three harmonics may be generated based on the first audio signal, such as the frequency range of the generated second harmonic signal includes: 20 Hz to 400 Hz, 40 Hz to 800 Hz, 60 Hz to 1200 Hz. The second harmonic signal is then filtered through the second filter group to obtain the first harmonic signal, that is, the virtual bass signal.

It can be understood that the second filter group is used to pass the signal that meets the frequency band range of the micro-speaker of the electronic device to selectively filter out the signal that does not meet the frequency band range of the micro-speaker of the electronic device, and the filters of the second filter group include high-pass filters and low-pass filters. The filters of the second filter group are linear filters, such as Bessel filters.

In the embodiment of the present application, the bandwidth of the second filter group can be determined according to the bandwidth of the micro speaker of the electronic device.

In some embodiments, the cutoff frequency of the low-pass filter in the second filter group can be determined according to the high-frequency cutoff frequency of the micro-speaker, and the cutoff frequency of the high-pass filter in the second filter group can be determined according to the low-frequency cutoff frequency of the micro-speaker. In other embodiments, the cutoff frequency of the low-pass filter in the second filter group can be determined according to the fundamental frequency of the audio signal to be reproduced, and the cutoff frequency of the high-pass filter in the second filter group can be determined according to the low-frequency cutoff frequency of the micro-speaker.

For example, the second filter group forms a bandpass of 400 Hz to 10 KHz, and the second filter group can filter the harmonic signals less than 400 Hz and the harmonic signals greater than 10 KHz in the second harmonic signal to obtain the first harmonic signal with a frequency range of 400 Hz to 10 KHz.

103: Perform feature recognition on the first audio signal to determine a signal type of the first audio signal.

In an embodiment of the present application, feature recognition is performed on the first audio signal to determine the signal type of the first audio signal; wherein, feature recognition of the first audio signal can be performed based on a support vector machine and preset features, and the preset features include time domain features, which have the characteristics of small computational complexity and can effectively distinguish between human voice and background sound. In some embodiments, the time domain features include zero crossing rate and waveform factor.

In some embodiments, the preset features include time domain features and frequency domain features, and the frequency domain features include frequency, spectrum, frequency bandwidth, frequency center, etc.

104: Determine a harmonic ratio of the first audio signal in real time according to the signal type of the first audio signal.

In the embodiment of the present application, the signal type corresponding to the first audio signal is a background sound signal, and the harmonic ratio of the first audio signal is reduced from a preset ratio to a first ratio. The signal type corresponding to the first audio signal is a voice signal, and the harmonic ratio of the first audio signal is reduced from a preset ratio to a second ratio, wherein the first ratio is less than the second ratio.

It is understandable that the harmonic content of an audio signal can affect the timbre and texture of the sound. The more harmonics an audio signal has, the richer and more complex it sounds, and the fewer harmonics an audio signal has, the purer it sounds.

It can be understood that the audio characteristics of the human voice signal and the background sound signal in the low-frequency signal are not exactly the same. For example, in general, the thickness of the human voice signal in the low-frequency signal is higher than that of the background sound signal. In order to take into account the audio effects of the background sound signal and the human voice signal in the low-frequency signal, such as in order to take into account the drum sound and the human voice effect in the low-frequency signal, the embodiment of the present application adopts a method of dynamically controlling the harmonic ratio, so that the decreasing speed of the harmonic ratio of the human voice signal is greater than the decreasing speed of the harmonic ratio of the background sound signal, that is, the first speed is less than the second speed, thereby improving the clarity of the human voice, while ensuring the low-frequency impact force and improving the clarity. For example, the signal type corresponding to the first audio signal is a background sound signal, such as drum sound, bass sound, etc., and the preset ratio is reduced at the first speed to obtain a first ratio; the signal type corresponding to the first audio signal is a human voice signal, and the preset ratio is reduced at the second speed to obtain a second ratio, thereby ensuring that both the harmonic signal and the fundamental signal can present corresponding audio effects.

105: Based on the harmonic ratio of the first audio signal, add the original signal of the first audio signal and the first harmonic signal to obtain a second audio signal.

In the embodiment of the present application, based on the harmonic ratio of the first audio signal, the original signal of the first audio signal and the first harmonic signal corresponding to the first audio signal can be added to obtain a second audio signal, and then the second audio signal can be played through the micro speaker.

In summary, in the embodiment of the present application, a Bessel filter is used to filter the audio signal. Since the Bessel filter is a linear filter, the audio signal filtered by the linear filter has a linear phase, which ensures that the relative phase relationship of each frequency component of the filtered audio signal does not change, thereby avoiding the problem of phase mismatch between the audio signal input after the filtered audio signal, that is, avoiding the problem of phase mismatch between the virtual bass signal and the original audio signal.

In addition, the embodiment of the present application utilizes a machine learning method to identify components in the first audio signal, and adjusts the harmonic ratio in real time according to the signal type of the first audio signal, so that the obtained signal with a low-frequency hearing sense not only retains the hearing sense corresponding to the high-frequency signal in the input original audio signal, but also can virtualize the bass hearing sense of the low-frequency signal in the input original audio signal, so that the user can intuitively feel the low-frequency hearing sense of the low-frequency audio, thereby improving the playback effect of the audio signal.

Fig. 4 shows a schematic diagram of an audio processing method according to an embodiment of the present application. It can be understood that the execution subject of the schematic diagram shown in Fig. 4 is the electronic device 100. In order to simplify the description, the execution subject of each step will not be repeatedly described below when introducing the schematic diagram shown in Fig. 4.

As shown in FIG4 , the audio processing method includes: the electronic device obtains the original audio signal of the input (Din), and divides the original audio signal into two channels, the original audio signal of the first channel is input into the Bessel filter group 1, the original audio signal is filtered by the Bessel filter group 1 (or referred to as the first filter group), and the filtered first audio signal is obtained; the first audio signal is feature-recognized by the digital signal processor to determine the signal type of the first audio signal. The second harmonic signal is generated based on the cutoff frequency of the micro speaker of the electronic device and the frequency of the first audio signal by a bandpass filter or a harmonic exciter; the second harmonic signal is filtered by the Bessel filter group 2 (or referred to as the second filter group) to obtain the filtered first harmonic signal that meets the bandwidth of the electronic device speaker, that is, the virtual bass signal. Corresponding to determining that the first audio signal is a background sound signal, such as drum sounds, bass sounds, etc., the preset proportion is reduced at a first speed, so that the harmonic proportion of the first audio signal is reduced from the preset proportion to the first proportion; corresponding to determining that the first audio signal is a human voice signal, the preset proportion is reduced at a second speed, so that the harmonic proportion of the first audio signal is reduced from the preset proportion to the second proportion, wherein the preset proportion is the ratio of the harmonic component to the fundamental wave component, the first proportion is less than the second proportion, and the first speed is greater than the second speed.

The original audio signal of the second channel is input into the adder, and when the first harmonic signal and the harmonic ratio of the first audio signal are generated in the first channel, the first harmonic signal is added to the original audio signal of the second channel based on the harmonic ratio of the first audio signal to obtain an output (Dout) audio signal.

In some embodiments, after the first channel outputs the first harmonic signal and the harmonic ratio of the first audio signal, the original audio signal of the second channel is input into the adder. In other embodiments, the original audio signal of the second channel may be input into the adder first, and then the first harmonic signal output by the first channel of the first channel and the harmonic ratio of the first audio signal are input into the adder. In other embodiments, the first harmonic signal output by the first channel, the harmonic ratio of the first audio signal, and the original audio signal of the second channel may be input into the adder at the same time.

In an embodiment of the present application, a Bessel filter is used to filter the audio signal. Since the Bessel filter is a linear filter, the audio signal filtered by the linear filter has a linear phase, which ensures that the relative phase relationship of each frequency component in the signal does not change, thereby avoiding the problem of phase mismatch between the virtual bass signal and the input audio signal.

An embodiment of the present application provides a chip, including a circuit, and the circuit is used to execute the above audio processing method.

This embodiment is an implementation of a chip corresponding to the above-mentioned audio processing method embodiment, and this embodiment can be implemented in conjunction with the above-mentioned audio processing method implementation. The relevant technical details mentioned in the above-mentioned audio processing method implementation are still valid in this embodiment, and in order to reduce repetition, they are not repeated here. Accordingly, the relevant technical details mentioned in this embodiment can also be applied in the circuit implementation.

An embodiment of the present application provides an electronic device, characterized in that it includes the above-mentioned chip.

This embodiment is an implementation of an electronic device corresponding to the above chip embodiment, and this embodiment can be implemented in conjunction with the above audio processing method implementation. The relevant technical details mentioned in the above audio processing method implementation are still valid in this embodiment, and in order to reduce repetition, they are not repeated here. Accordingly, the relevant technical details mentioned in this embodiment can also be applied in the circuit implementation.

According to an embodiment of the present application, FIG5 shows a block diagram of an electronic device 1300 based on a system on chip (SOC). In FIG5, similar components have the same reference numerals. In addition, the dashed box is an optional feature of a more advanced SOC. In FIG5, the electronic device 1300 includes: an interconnect unit 1350, which is coupled to a processor 1315, and the processor 1315 can be used to control the electronic device to perform the audio processing method of the present application; a system agent unit 1370; a bus controller unit 1380; an integrated memory controller unit 1340; a group or one or more coprocessors 1320, which may include a first filter group, a second filter group, an adder, an audio processor, and a video processor; a static random access memory (SRAM) unit 1330; a direct memory access (DMA) unit 1360. In one embodiment, the coprocessor 1320 includes a dedicated processor, such as, for example, a network or communication processor, a compression engine, or an embedded processor.

In some cases, the disclosed embodiments may be implemented in hardware, firmware, software, or any combination thereof. The disclosed embodiments may also be implemented as instructions carried or stored on one or more temporary or non-temporary machine-readable (e.g., computer-readable) storage media, which may be read and executed by one or more processors. For example, instructions may be distributed over a network or through other computer-readable media. Therefore, machine-readable media may include any mechanism for storing or transmitting information in a machine (e.g., computer) readable form, including but not limited to floppy disks, optical disks, optical disks, read-only memories (compact disc-read only memories, CD-ROMs), magneto-optical disks, read-only memories (read only memory, ROM), random access memories (random access memory, RAM), erasable programmable read-only memories (erasable programmable read-only memories, EPROM), electrically erasable programmable read-only memories (electrically erasable programmable read-only memories, EEPROM), magnetic or optical cards, flash memory, or a tangible machine-readable memory for transmitting information (e.g., carrier waves, infrared signals, digital signals, etc.) using the Internet in electrical, optical, acoustic, or other forms of propagation signals. Accordingly, machine-readable media include any type of machine-readable media suitable for storing or transmitting electronic instructions or information in a form readable by a machine (eg, a computer).

In the accompanying drawings, some structural or method features may be shown in a specific arrangement and/or order. However, it should be understood that such a specific arrangement and/or order may not be required. Instead, in some embodiments, these features may be arranged in a manner and/or order different from that shown in the illustrative drawings. In addition, the inclusion of structural or method features in a particular figure does not mean that such features are required in all embodiments, and in some embodiments, these features may not be included or may be combined with other features.

It should be noted that the units/modules mentioned in the various device embodiments of the present application are all logical units/modules. Physically, a logical unit/module can be a physical unit/module, or a part of a physical unit/module, or can be implemented as a combination of multiple physical units/modules. The physical implementation method of these logical units/modules themselves is not the most important. The combination of functions implemented by these logical units/modules is the key to solving the technical problems proposed by the present application. In addition, in order to highlight the innovative part of the present application, the above-mentioned device embodiments of the present application do not introduce units/modules that are not closely related to solving the technical problems proposed by the present application, which does not mean that there are no other units/modules in the above-mentioned device embodiments.

It should be noted that in the examples and description of this patent, the terms "include", "comprises" or any other variants thereof are intended to cover non-exclusive inclusion, so that a process, method, article or device including a series of elements includes not only those elements, but also includes other elements not explicitly listed, or also includes elements inherent to such process, method, article or device. In the absence of further restrictions, an element defined by the sentence "includes one" does not exclude the presence of other identical elements in the process, method, article or device including the element. Although the present application has been illustrated and described with reference to certain preferred embodiments of the present application, it should be understood by those skilled in the art that various changes may be made to it in form and detail without departing from the spirit and scope of the present application.

Claims (12)

1. An audio processing method, characterized in that it is applied to an electronic device, the audio processing method comprising: Acquire a first audio signal; generating a first harmonic signal based on the first audio signal; performing feature recognition on the first audio signal to determine a signal type of the first audio signal; determining a harmonic ratio of the first audio signal in real time according to a signal type of the first audio signal; Based on the harmonic ratio of the first audio signal, an original audio signal of the first audio signal and the first harmonic signal are added together to obtain a second audio signal. 2. The audio processing method according to claim 1, wherein determining the harmonic ratio of the first audio signal in real time according to the signal type of the first audio signal comprises: Corresponding to the signal type of the first audio signal being a background sound signal, reducing the harmonic ratio of the first audio signal from a preset ratio to a first ratio; Corresponding to the signal type of the first audio signal being a human voice signal, the harmonic ratio of the first audio signal is reduced from a preset ratio to a second ratio, wherein the first ratio is smaller than the second ratio. 3. The audio processing method according to claim 2, wherein reducing the harmonic ratio of the first audio signal from a preset ratio to a first ratio comprises: reducing the harmonic ratio of the first audio signal from the preset ratio to the first ratio based on a first speed; The step of reducing the harmonic ratio of the first audio signal from a preset ratio to a second ratio includes: reducing the harmonic ratio of the first audio signal from the preset ratio to the second ratio based on a second speed; Wherein, the first speed is lower than the second speed. 4. The audio processing method according to claim 1, wherein the performing feature recognition on the first audio signal to determine the signal type of the first audio signal comprises: Feature recognition is performed on the first audio signal based on preset features to determine a signal type of the first audio signal, wherein the preset features include a zero crossing rate and a waveform factor. 5. The audio processing method according to claim 1, wherein obtaining the first audio signal comprises: Get the original audio signal; Filter the original audio signal through a first filter group to obtain the first audio signal; The generating a first harmonic signal based on the first audio signal comprises: generating a second harmonic signal based on the first audio signal; The second harmonic signal is filtered through a second filter group to obtain the first harmonic signal. 6 . The audio processing method according to claim 5 , wherein the first filter group and the second filter group both comprise a high-pass filter and a low-pass filter, and the high-pass filter and the low-pass filter are both linear filters. 7 . The audio processing method according to claim 6 , wherein the linear filter comprises a Bessel filter. 8. The audio processing method according to any one of claims 5 to 6 is characterized in that the bandwidth of the first filter group is determined based on the audio bandwidth of the original audio signal and the bandwidth of the speaker of the electronic device; the bandwidth of the second filter group is determined based on the bandwidth of the speaker of the electronic device. 9. The audio processing method according to any one of claims 5 to 6, characterized in that the bandwidth of the first filter group is determined based on the audio bandwidth of the original audio signal and the bandwidth of the speaker of the electronic device, comprising: The cutoff frequency of the high-pass filter in the first filter group is determined based on the lower limit frequency of the original audio signal and the low-frequency cutoff frequency of the speaker of the electronic device, and the cutoff frequency of the low-pass filter in the first filter group is determined based on the low-frequency cutoff frequency and the high-frequency cutoff frequency of the speaker of the electronic device. 10. The audio processing method according to any one of claims 5 to 6, characterized in that the bandwidth of the second filter group is determined based on the bandwidth of the speaker of the electronic device, comprising: The cutoff frequency of the high-pass filter in the second filter group is determined based on the low-frequency cutoff frequency of the speaker of the electronic device, and the cutoff frequency of the low-pass filter in the second filter group is determined based on the high-frequency cutoff frequency of the speaker of the electronic device. 11. A chip, characterized by comprising a circuit, wherein the circuit is used to execute the audio processing method according to any one of claims 1 to 10. 12. An electronic device, comprising the chip according to claim 11.