CN118233809A - Audio signal processing method, device, electronic device and storage medium - Google Patents

Abstract

The invention provides an audio signal processing method, an audio signal processing device, electronic equipment and a storage medium, wherein the method comprises the following steps: acquiring an audio signal after mixing left and right channels; adjusting the magnitude of the level value in the audio signal based on the digital signal processing; based on the magnitude of the level value, carrying out dynamic bass enhancement processing on the audio signals corresponding to the left and right channels; respectively carrying out superposition processing on the processed audio signals of the left channel and the right channel and the original audio signals of the corresponding channels; and outputting the audio signals of the left channel and the right channel after the superposition processing. The invention adjusts the level value of the audio signal after the left and right channels are mixed based on digital signal processing, and carries out dynamic bass enhancement processing on the audio signal of the left and right channels after adjustment, converts the audio signal into corresponding control signals to control the gains of two paths of extracted audio signals, and mixes the two paths of extracted audio signals with the original audio signal of the left and right channels, thereby achieving multi-frequency band dynamic compensation of the audio signal and realizing low-cost dynamic effect.

Description

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

Technical Field

The present invention relates to signal processing technology, and in particular to an audio signal processing method, device, electronic device and computer-readable storage medium.

Background technique

With the rapid development of the automobile industry and the continuous improvement of people's living standards, cars have gradually become an indispensable means of transportation in people's daily lives. With the increasing number of automobile brands, consumers are increasingly demanding on the quality of their own consumption. Therefore, many automobile companies have begun to focus on the driving experience and comfort in the car. In addition to the hardware configuration of the vehicle itself, the sound effect in the car is also a major part of improving the comfort of the vehicle.

In the related technology, in order to improve the sound quality and sound effects in the car, a large amount of money needs to be invested in the research and development or purchase of sound effect algorithms, which undoubtedly increases the cost.

Therefore, how to effectively improve the sound quality inside the car at a low cost is a technical problem that needs to be solved.

Summary of the invention

The present invention provides an audio signal processing method, device, electronic device and computer-readable storage medium to at least solve the problem of increased costs due to the need to spend a lot of money to improve the sound quality in the car in the related art. The technical solution of the present invention is as follows:

According to a first aspect of an embodiment of the present invention, there is provided an audio signal processing method, comprising:

Get the audio signal after mixing the left and right channels;

Adjusting the level value of the audio signal based on digital signal processing;

Based on the adjusted level value, performing dynamic bass enhancement processing on the audio signals corresponding to the left and right channels;

Superimposing the processed audio signals of the left and right channels with the original audio signals of the corresponding channels respectively;

Output the audio signals of the left and right channels after superposition processing.

Optionally, adjusting the level value of the audio signal based on digital signal processing includes:

Detecting the level value of the bass volume in the audio signal according to the response time and release time of the level based on digital signal processing;

The level value is adjusted according to different coefficient transformations.

Optionally, adjusting the level value according to different coefficient transformations includes:

The level value is multiplied by different set coefficients to adjust the level value of the bass volume in the audio signal.

Optionally, the performing dynamic bass enhancement processing on the audio signals corresponding to the left and right channels based on the adjusted level value includes:

Adjusting the gains of the audio signals corresponding to the left and right channels based on the adjusted level values;

The audio signals of the left and right channels after gain adjustment are input into corresponding filters for compensation processing of different frequency bands.

Optionally, adjusting the gain of the audio signal corresponding to the left and right channels based on the adjusted level value includes: multiplying the adjusted level value by a dynamic factor, and adjusting the dynamic gain of volume adjustment in the audio signal corresponding to the left and right channels based on the multiplication result;

The step of inputting the gain-adjusted audio signals of the left and right channels into corresponding filters for compensation processing of different frequency bands includes: inputting the gain-adjusted audio signals of the left and right channels into corresponding low-pass filters and/or high-pass filters, respectively, for compensation processing of different frequency bands; or, inputting the gain-adjusted audio signals of the left and right channels into corresponding one or more band-pass filters for multi-stage compensation processing of different frequency bands.

Optionally, the method further includes:

Obtain the audio signals of the left and right channels within a set time period after superposition processing;

Performing calculation processing on the obtained audio signals of the left and right channels respectively to obtain the phase difference and center channel signal of the corresponding channels after processing;

Performing virtual surround sound processing on the processed phase difference and center channel signal respectively to obtain a left channel audio signal after sound image localization and a right channel audio signal after mid-low frequency compensation;

The right channel audio signal after mid-low frequency compensation, the left channel audio signal after sound image localization, and the corresponding left and right channel original audio signals are respectively mixed, and the left channel audio signal after left mixing processing and the right channel audio signal after right mixing processing are output.

Optionally, performing operation processing on the acquired audio signals of the left and right channels respectively to obtain the phase difference of the corresponding channels and the center channel signal after processing includes:

Subtracting the acquired audio signals of the left and right channels to obtain a phase difference of the audio signal within the set time period;

The acquired audio signals of the left and right channels are added to simulate a center channel signal within the set time period.

Optionally, the processing of the processed phase difference and center channel signal respectively for virtual surround sound to obtain a left channel audio signal after sound image localization and a right channel audio signal after mid-low frequency compensation includes:

Performing frequency compensation on the phase difference based on the conclusion of auditory spatial localization mechanism to obtain a right channel audio signal after mid-low frequency compensation;

The center audio signal is frequency controlled by a central sound control mechanism to obtain a left channel audio signal after sound image localization.

Optionally, the frequency compensation of the phase difference is performed based on the conclusion of the auditory space localization mechanism to obtain a right channel audio signal after mid- and low-frequency compensation, including:

The phase difference is frequency compensated by using a Peaking filter based on the conclusion of the auditory spatial localization mechanism to obtain a right channel audio signal after mid- and low-frequency compensation.

According to a second aspect of an embodiment of the present invention, a method for processing an audio signal is provided, the method comprising:

Acquire audio signals of left and right channels within a set time period, wherein the audio signals include: audio signals of left and right channels before mixing the left and right channels; or audio signals of left and right channels after output superposition processing;

The obtained audio signals of the left and right channels are processed respectively to obtain the phase difference and center channel signal of the corresponding channels after processing;

Performing virtual surround sound processing on the processed phase difference and center channel signal respectively to obtain a left channel audio signal after sound image localization and a right channel audio signal after mid-low frequency compensation;

Perform left-right mixing processing on the right channel audio signal after mid-low frequency compensation, the left channel audio signal after sound image localization, and the corresponding left and right channel original audio signals respectively;

Output the left channel audio signal after left mixing processing and the right channel audio signal after right mixing processing.

According to a third aspect of an embodiment of the present invention, there is provided an audio signal processing device, the device comprising:

A first acquisition module is used to obtain an audio signal after the left and right channels are mixed;

An adjustment module, used for adjusting the level value of the audio signal based on digital signal processing;

A processing module, configured to perform dynamic bass enhancement processing on the audio signals corresponding to the left and right channels based on the adjusted level value;

A superposition module, used for superimposing the processed audio signals of the left and right channels with the original audio signals of the corresponding channels respectively;

The output module is used to output the audio signals of the left and right channels after superposition processing.

Optionally, the adjustment module includes:

A detection module, configured to detect a level value of a bass volume in the audio signal according to a response time and a release time of the level based on digital signal processing;

The level adjustment module is used to adjust the level value according to different coefficient transformations.

Optionally, the level adjustment module is specifically configured to multiply the level value by different set coefficients to adjust the level value of the bass volume in the audio signal.

Optionally, the processing module includes:

A gain adjustment module, used for adjusting the gain of the audio signal corresponding to the left and right channels based on the adjusted level value;

The compensation processing module is used to input the audio signals of the left and right channels after gain adjustment into corresponding filters for compensation processing of different frequency bands.

Optionally, the gain adjustment module is specifically configured to multiply the adjusted level value by a dynamic factor, and adjust the dynamic gain of volume adjustment in the audio signals corresponding to the left and right channels based on the multiplication result;

The compensation processing module is specifically used to input the audio signals of the left and right channels after adjusting the gains into corresponding low-pass filters and/or high-pass filters respectively for compensation processing in different frequency bands; or, input the audio signals of the left and right channels after adjusting the gains into corresponding one or more band-pass filters for multi-stage compensation processing in different frequency bands.

Optionally, the device further comprises:

The second acquisition module is used to acquire the audio signals of the left and right channels within a set time period after superposition processing;

An operation processing module, used to perform operation processing on the acquired audio signals of the left and right channels respectively, to obtain the phase difference and center channel signal of the corresponding channels after processing;

A virtual surround processing module, used to perform virtual surround sound processing on the processed phase difference and center channel signal respectively, to obtain a left channel audio signal after sound image localization and a right channel audio signal after mid-low frequency compensation;

The mixing processing module is used to perform left-right mixing processing on the right channel audio signal after mid-low frequency compensation, the left channel audio signal after sound image localization, and the corresponding left and right channel original audio signals, and output the left channel audio signal after left mixing processing and the right channel audio signal after right mixing processing.

Optionally, the operation processing includes:

A subtraction processing module, used to subtract the acquired audio signals of the left and right channels to obtain a phase difference of the audio signal within the set time period;

The addition processing module is used to add the acquired audio signals of the left and right channels to simulate the center channel signal within the set time period.

Optionally, the virtual surround processing module includes:

A frequency compensation processing module, used to perform frequency compensation on the phase difference according to the conclusion of the auditory space positioning mechanism, and obtain a right channel audio signal after mid-low frequency compensation;

The frequency control processing module is used to perform frequency control on the center audio signal through a central sound control mechanism to obtain a left channel audio signal after sound image localization.

Optionally, the frequency compensation processing module is specifically used to perform frequency compensation on the phase difference using a Peaking filter based on the conclusion of the auditory space localization mechanism to obtain a right channel audio signal after mid- and low-frequency compensation.

According to a fourth aspect of an embodiment of the present invention, there is provided an audio signal processing device, the device comprising:

An acquisition module, used to acquire audio signals of left and right channels within a set time period, wherein the audio signals include: audio signals of left and right channels before mixing the left and right channels; or audio signals of left and right channels after output superposition processing;

An operation processing module, used to perform operation processing on the acquired audio signals of the left and right channels respectively, to obtain the phase difference and center channel signal of the corresponding channels after processing;

A virtual sound processing module, used for performing virtual surround sound processing on the processed phase difference and center channel signals respectively, to obtain a left channel audio signal after sound image localization and a right channel audio signal after mid-low frequency compensation;

A mixing processing module, used for performing left-right mixing processing on the right channel audio signal after mid-low frequency compensation, the left channel audio signal after sound image localization, and the corresponding left and right channel original audio signals respectively;

The output module is used to output the left channel audio signal after the left mixing process and the right channel audio signal after the right mixing process.

Optionally, the operation processing includes:

A subtraction processing module, used to subtract the acquired audio signals of the left and right channels to obtain a phase difference of the audio signal within the set time period;

The addition processing module is used to add the acquired audio signals of the left and right channels to simulate the center channel signal within the set time period.

Optionally, the virtual surround processing module includes:

A frequency compensation processing module, used to perform frequency compensation on the phase difference according to the conclusion of the auditory space positioning mechanism, and obtain a right channel audio signal after mid-low frequency compensation;

The frequency control processing module is used to perform frequency control on the center audio signal through a central sound control mechanism to obtain a left channel audio signal after sound image localization.

Optionally, the frequency compensation processing module is specifically used to perform frequency compensation on the phase difference using a Peaking filter based on the conclusion of the auditory space localization mechanism to obtain a right channel audio signal after mid- and low-frequency compensation.

According to a fifth aspect of an embodiment of the present invention, there is provided an electronic device, including:

processor;

a memory for storing instructions executable by the processor;

The processor is configured to execute the instructions to implement the audio signal processing method as described above.

According to a sixth aspect of an embodiment of the present invention, a computer-readable storage medium is provided. When instructions in the computer-readable storage medium are executed by a processor of an electronic device, the electronic device is enabled to execute the method for processing an audio signal as described above.

According to a seventh aspect of an embodiment of the present invention, there is provided a computer program product, comprising a computer program or instructions, wherein when the computer program or instructions are executed by a processor of an electronic device, the method for processing an audio signal as described above is implemented.

The technical solution provided by the embodiments of the present invention brings at least the following beneficial effects:

In an embodiment of the present invention, an audio signal after mixing left and right channels is obtained; the level value in the audio signal is adjusted based on digital signal processing; based on the adjusted level value, the audio signals corresponding to the left and right channels are dynamically enhanced by bass; the processed audio signals of the left and right channels are respectively superimposed with the original audio signals of the corresponding channels; and the audio signals of the left and right channels after superimposition are output. That is to say, in an embodiment of the present invention, the level value of the audio signal after mixing left and right channels is adjusted based on digital signal processing, and the audio signals corresponding to the adjusted left and right channels are dynamically enhanced by bass, and converted into corresponding control signals to control the gain of the two audio signals derived, and then the two audio signals are mixed with the original audio signals of the left and right channels, so as to achieve multi-band dynamic compensation of the audio signal and realize low-cost dynamic bass effect.

It is to be understood that both the foregoing general description and the following detailed description are exemplary and explanatory only and are not restrictive of the invention.

BRIEF DESCRIPTION OF THE DRAWINGS

The drawings herein are incorporated into the specification and constitute a part of the specification, showing embodiments consistent with the present invention, and are used together with the specification to explain the principles of the present invention, and do not constitute an improper limitation of the present invention. In order to more clearly illustrate the technical solutions in the embodiments of the present application or the prior art, the drawings required for use in the embodiments or the prior art description will be briefly introduced below. Obviously, the drawings described below are some embodiments of the present application. For ordinary technicians in this field, other drawings can be obtained based on these drawings without paying creative work.

FIG. 1 is a flow chart of a method for processing an audio signal provided by an embodiment of the present invention.

FIG. 2 is another flow chart of a method for processing an audio signal provided by an embodiment of the present invention.

FIG. 3 is a schematic diagram of an application example of a method for processing an audio signal provided by an embodiment of the present invention.

FIG. 4 is another schematic diagram of an application example of an audio signal processing method provided by an embodiment of the present invention.

FIG. 5 is a schematic diagram of frequency compensation provided by an embodiment of the present invention.

FIG. 6 is a block diagram of an audio signal processing device provided by an embodiment of the present invention.

FIG. 7 is another block diagram of an audio signal processing device provided by an embodiment of the present invention.

FIG8 is a block diagram of an electronic device provided by an embodiment of the present invention.

FIG. 9 is a block diagram of a device for processing an audio signal provided by an embodiment of the present invention.

FIG. 10 is another block diagram of an electronic device provided by an embodiment of the present invention.

Detailed ways

In order to enable ordinary persons in the art to better understand the technical solutions of the present invention, the technical solutions in the embodiments of the present invention will be clearly and completely described below in conjunction with the accompanying drawings.

It should be noted that the terms "first", "second", etc. in the specification and claims of the present invention and the above-mentioned drawings are used to distinguish similar objects, and are not necessarily used to describe a specific order or sequence. It should be understood that the data used in this way can be interchanged where appropriate, so that the embodiments of the present invention described herein can be implemented in an order other than those illustrated or described herein. The embodiments described in the following exemplary embodiments do not represent all embodiments consistent with the present invention. Instead, they are merely examples of devices and methods consistent with some aspects of the present invention as detailed in the appended claims.

FIG. 1 is a flow chart of a method for processing an audio signal provided by an embodiment of the present invention. As shown in FIG. 1 , the method for processing an audio signal includes the following steps:

Step 101: Obtaining a mixed audio signal of left and right channels;

Step 102: adjusting the level value of the audio signal based on digital signal processing;

Step 103: Based on the adjusted level value, dynamically enhance the audio signals corresponding to the left and right channels with bass enhancement;

Step 104: superimposing the processed audio signals of the left and right channels with the original audio signals of the corresponding channels respectively;

Step 105: outputting the audio signals of the left and right channels after the superposition processing.

In an embodiment of the present invention, an audio signal after mixing left and right channels is obtained; the level value in the audio signal is adjusted based on digital signal processing; based on the adjusted level value, dynamic bass enhancement is performed on the audio signals corresponding to the left and right channels; the processed audio signals of the left and right channels are respectively superimposed with the original audio signals of the corresponding channels; and the audio signals of the left and right channels after superimposition are output. That is to say, in an embodiment of the present invention, the level value of the audio signal after mixing left and right channels is adjusted based on digital signal processing, and the adjusted audio signals of the left and right channels are dynamically enhanced, which are converted into corresponding control signals to control the gain of the two audio signals derived, and then mixed with the original audio signals of the left and right channels, thereby achieving multi-band dynamic compensation of the audio signal and realizing low-cost dynamic effects.

The audio signal processing method described in the present invention can be applied to terminals, servers, vehicle terminals, etc., without limitation. The terminal implementation device can be electronic devices such as smart phones, laptops, tablet computers, desktop computers, personal digital assistants (PDA, Personal Digital Assistant) and wearable devices. The vehicle terminal can be electronic devices such as vehicle terminals, vehicle control platforms, industrial computers, etc. The server can be an independent server, or a server cluster, or a server that provides cloud services, cloud databases, cloud computing, cloud functions, cloud storage, network services, cloud communications, intermediate services, domain name services, security services, content distribution networks, or big data and artificial intelligence platforms, etc., without limitation.

1 , specific implementation steps of an audio signal processing method provided by an embodiment of the present invention are described in detail below.

In step 101, an audio signal after mixing left and right channels is obtained.

In this step, the electronic device first obtains the audio signals in the left and right channels to be played, and then mixes the obtained left and right audio signals to obtain a mixed audio signal. The mixing method is to superimpose the left and right audio signals.

In step 102, the level value of the audio signal is adjusted based on digital signal processing.

In this step, the digital signal processing DSP detects the level value of the bass volume in the audio signal according to the response time and release time of the level, and then adjusts the level value according to different coefficient transformations.

Specifically, in this embodiment, based on digital signal processing (DSP, Digital Signal Process), only the level value of the low-frequency signal in the audio signal can be monitored, usually according to the response time and release time of the level, so as to change the real-time and smoothness of the signal detection. Usually, the envelope signal of the positive half cycle of the signal is taken. It should be noted that the envelope signal in the signal refers to a high-frequency amplitude modulation signal. Its amplitude changes according to the low-frequency modulation signal. If the peak points of the high-frequency modulation signal are connected, a curve corresponding to the low-frequency modulation signal can be obtained. This curve is the envelope line. It is two curves that can wrap the signal waveform.

The adjusting the level value by transforming according to different coefficients includes: multiplying the level value by different set coefficients to adjust the level value of the low volume in the audio signal.

In this embodiment, different coefficients are preset, that is, the detected level value is multiplied by different preset coefficients respectively to obtain level values of different sizes, thereby changing the size of the level value in turn.

In step 103, based on the adjusted level value, dynamic bass enhancement processing is performed on the audio signals corresponding to the left and right channels.

In this step, based on the adjusted level value, the gains of the audio signals corresponding to the left and right channels are adjusted; and the audio signals of the left and right channels after the gains are adjusted are input into corresponding filters for compensation processing of different frequency bands.

Wherein, adjusting the gain of the audio signal corresponding to the left and right channels based on the adjusted level value comprises: multiplying the adjusted level value by a dynamic factor, and adjusting the dynamic gain of the volume adjustment in the audio signal corresponding to the left and right channels based on the multiplication result; and

The step of inputting the gain-adjusted audio signals of the left and right channels into corresponding filters for compensation processing of different frequency bands includes: inputting the gain-adjusted audio signals of the left and right channels into corresponding low-pass filters and/or high-pass filters, respectively, for compensation processing of different frequency bands; or, inputting the gain-adjusted audio signals of the left and right channels into corresponding one or more band-pass filters for multi-stage compensation processing of different frequency bands.

In this embodiment, the value of the subsequent gain can be changed according to the magnitude of the detected level value, specifically by setting coefficients of different frequency bands for the transformation.

Specifically, when the level value is greater than the set threshold, the subsequent gain is 0, that is, the original audio signal is not processed and is directly output. When the level value is less than or equal to the set threshold, the level value is multiplied by different dynamic factors, and the dynamic gain of the volume adjustment in the audio signal corresponding to the left and right channels is adjusted based on the multiplication result. The smaller the level value, the greater its gain. Afterwards, the audio signals of the left and right channels after the gain is adjusted are respectively input into the corresponding low-pass filter and/or high-pass filter for compensation processing of different frequency bands; or, the audio signals of the left and right channels after the gain is adjusted are input into the corresponding one or more band-pass filters for multi-stage compensation processing of different frequency bands.

It should be noted that for the same input level, the gain control obtained after the coefficients in different frequency bands are transformed is also different. For example, the gain of the low-pass band is generally larger than that of the high-pass band, which is obtained from the equal loudness curve.

Among them, the multi-stage compensation part in this embodiment is realized by a bandpass filter, which can be expanded into multiple bandpass filters as needed. In this way, users can input audio signals of different frequency bands into filters of different frequency bands for filtering processing as needed, thereby achieving the effect of multi-stage compensation.

In step 104, the processed audio signals of the left and right channels are respectively superimposed with the original audio signals of the corresponding channels.

In this step, the processed audio signal of the left channel is added to the original audio signal of the corresponding left channel, so as to obtain the added left channel audio signal.

Similarly, the processed audio signal of the right channel is added to the original audio signal of the corresponding right channel to obtain an added right channel audio signal.

In this step, the audio signals are filtered through different filters and then mixed with the original audio signals of the corresponding channels, thereby achieving the effect of multi-band dynamic compensation.

In step 105, the audio signals of the left and right channels after the superposition processing are output.

In this step, the audio signal of the left channel after the superposition processing is output through the left channel, and the audio signal of the right channel after the superposition processing is output through the right channel.

In the embodiment of the present invention, the level value of the audio signal after the left and right channels are mixed is adjusted based on digital signal processing, and the level value is converted into a corresponding control signal to control the gain of the two-way audio signal. After the two-way audio signal after gain is processed with dynamic bass enhancement, it is mixed with the original left and right channel audio signals, thereby achieving multi-band dynamic compensation of the audio signal and realizing a low-cost dynamic bass effect.

Optionally, in another embodiment, based on the above embodiment, the method may also include: obtaining audio signals of left and right channels within a set time period after superposition processing; performing calculation processing on the obtained audio signals of left and right channels respectively to obtain the phase difference and center channel signal of the corresponding channels after processing; performing virtual surround sound processing on the processed phase difference and center channel signal respectively to obtain the left channel audio signal after sound image positioning and the right channel audio signal after mid-low frequency compensation; performing left-right mixing processing on the right channel audio signal after mid-low frequency compensation and the left channel audio signal after sound image positioning with the corresponding original audio signals of left and right channels, and outputting the left channel audio signal after the left mixing processing and the right channel audio signal after the right mixing processing.

In an embodiment of the present invention, the audio signals of the left and right channels can be obtained and subjected to dynamic bass enhancement processing in the digital signal processing DSP, that is, based on the size of the adjusted level value, the audio signals corresponding to the left and right channels are subjected to dynamic bass enhancement processing, and the compensation amplitude can be determined by the size of the actual signal, thereby achieving a multi-level compensation effect; thereafter, the compensated audio signal is subjected to virtual surround sound processing, that is, through the auditory space positioning mechanism, the part with phase difference in the mid-low audio signal is extracted for frequency compensation, thereby achieving a low-cost dynamic bass effect.

Optionally, the acquired audio signals of the left and right channels are respectively processed to obtain the phase difference and center channel signal of the corresponding channels after processing, including: subtracting the acquired audio signals of the left and right channels to obtain the phase difference of the center audio signal within the set time period; adding the acquired audio signals of the left and right channels to simulate the center channel signal within the set time period.

Optionally, the processed phase difference and center channel signal are respectively subjected to virtual surround sound processing to obtain a left channel audio signal after sound image positioning and a right channel audio signal after mid- and low-frequency compensation, including: performing frequency compensation on the phase difference according to the conclusion of the auditory space positioning mechanism to obtain a right channel audio signal after mid- and low-frequency compensation; and performing frequency control on the center audio signal through a central sound control mechanism to obtain a left channel audio signal after sound image positioning.

In this embodiment, the sound image positioning and the effect of the front sound field can be improved by setting the central sound control, and the spatial sense and movement sense of the surround sound can be improved by setting the spatial sound control.

Among them, the frequency compensation of the phase difference is performed according to the conclusion of the auditory space positioning mechanism to obtain the right channel audio signal after mid-low frequency compensation, including: using the Peaking filter to frequency compensate the phase difference according to the conclusion of the auditory space positioning mechanism to obtain the right channel audio signal after mid-low frequency compensation.

In an embodiment of the present invention, based on the above embodiment, the audio signals of the left and right channels within a set time period after superposition processing are obtained, and then the phase difference of the set time period is extracted by subtracting the audio signals of the left and right channels, and then the frequency compensation of the mid-low frequency is performed through the auditory space positioning mechanism to increase the spatial positioning ability of the mid-low frequency, and simulate the virtual surround channel. By changing the level of the virtual surround channel, the ambient sound and various emitted sounds can be highlighted, and the sense of space and movement can be improved. And the audio signals of the left and right channels are added, and the center channel is simulated, and the sound image positioning and the front sound field effect of the center channel are changed. The specific implementation process is detailed in Figure 2 below, which will not be repeated here.

Please also refer to FIG. 2, which is another flow chart of a method for processing an audio signal provided by an embodiment of the present invention. The method includes:

Step 201: obtaining audio signals of left and right channels within a set time period; the audio signals include: audio signals of left and right channels before mixing; or audio signals of left and right channels after output superposition processing.

In this step, the audio signals of the left and right channels within a period of time are obtained. Specifically, they can be the audio signals of the left and right channels before the left and right channels are mixed in the embodiment of Figure 1; they can also be the audio signals of the left and right channels after the output superposition processing, etc., which is not limited in this embodiment.

Step 202: performing calculation processing on the acquired audio signals of the left and right channels respectively to obtain the phase difference and center channel signal of the corresponding channels after processing.

In this step, the obtained audio signals of the left and right channels are subtracted to obtain the phase difference of the center audio signal within the set time period; and the obtained audio signals of the left and right channels are added to simulate the center channel signal within the set time period.

Specifically, the audio signal of the right channel is input into the inverter to obtain an inverted audio signal of the right channel, and then the inverted audio signal of the right channel is added (actually subtracted) with the audio signal of the left channel to obtain the phase difference of the center audio signal within the set time period. And the obtained audio signals of the left and right channels are directly added to simulate the center channel signal within the set time period.

Step 203: performing virtual surround sound processing on the processed phase difference and center channel signal respectively to obtain a left channel audio signal after sound image localization and a right channel audio signal after mid-low frequency compensation.

In this step, the phase difference can be frequency compensated according to the conclusion of the auditory space localization mechanism to obtain the right channel audio signal after mid-low frequency compensation; specifically, the phase difference can be frequency compensated using a Peaking filter according to the conclusion of the auditory space localization mechanism to obtain the right channel audio signal after mid-low frequency compensation. And the center audio signal can be frequency controlled by the central sound control mechanism to obtain the left channel audio signal after sound image localization.

In this step, the right channel audio signal after frequency compensation to obtain mid-low frequency compensation is input to the inverter for signal inversion processing, and the left channel audio signal after sound image localization is input to the amplifier for signal amplification processing.

Step 204: Perform left-right mixing processing on the right channel audio signal after mid-low frequency compensation, the left channel audio signal after sound image localization, and the corresponding left and right channel original audio signals, and output the left channel audio signal after left mixing processing and the right channel audio signal after right mixing processing.

In this step, the left mixer can mix the right channel audio signal after mid-low frequency compensation, the left channel audio signal after sound image localization, and the original audio signal of the left channel to obtain the left channel audio signal after left mixing. Similarly, the right channel audio signal after mid-low frequency compensation, the left channel audio signal after sound image localization, and the original audio signal of the right channel can be mixed by the right mixer to obtain the right channel audio signal after right mixing. The specific mixing process of the left mixer and the right mixer is already a well-known technology for those skilled in the art and will not be repeated here.

In an embodiment of the present invention, audio signals of left and right channels within a set time period are obtained, and the obtained audio signals of left and right channels are processed respectively to obtain phase differences and center channel signals of corresponding channels after processing; virtual surround sound processing is performed on the processed phase differences and center channel signals respectively to obtain left channel audio signals after sound image localization and right channel audio signals after mid-low frequency compensation; left and right mixing processing is performed on the right channel audio signals after mid-low frequency compensation and the left channel audio signals after sound image localization, and the corresponding left and right channel original audio signals respectively; and the left channel audio signals after the left mixing processing and the right channel audio signals after the right mixing processing are output. That is to say, in the embodiment of the present invention, the acquired audio signals of the left and right channels are first processed based on digital signal processing, and then the processed phase difference and center channel signal are respectively subjected to virtual surround sound processing, that is, the phase difference is localized through the auditory space mechanism, and the part with phase difference in the mid-low audio signal is extracted for frequency compensation, the spatial localization capability of the mid-low frequency is increased, the surround channel is simulated, the sense of space and movement is improved, and the center channel is simulated by adding the left and right channels, and the sound image localization of the center channel and the effect of the front sound field are changed, thereby achieving a low-cost dynamic bass effect.

In the embodiment of the present invention, based on the digital signal processing DSP, the audio signals of the left and right channels are compensated by dynamic bass and virtual surround sound, so that the effects of virtual sound and dynamic bass can be achieved at a low cost. Among them, the dynamic bass is improved from the original equal loudness curve, and the compensation amplitude can be determined by the size of the actual signal and can be expanded to multiple levels of compensation. The virtual surround sound is realized by extracting the part with phase difference in the audio signal through the auditory space positioning mechanism for frequency compensation. The specific implementation process is detailed as follows.

Please also refer to FIG3, which is a schematic diagram of an application example of an audio signal processing method provided by an embodiment of the present invention, wherein the method performs compensation processing on the audio signals of the left and right channels through dynamic bass. This embodiment takes the acquisition of the audio signals of the left channel L and the right channel R as an example. As shown in FIG3, it includes: multiple adders, level detection, multiple coefficient transformations, and amplifiers (such as triodes, etc.) and corresponding filters connected to each coefficient transformation, such as a low-pass filter (referred to as low-pass), a high-pass filter (referred to as high-pass), and a band-pass filter (referred to as band-pass), etc., so as to achieve multi-stage compensation. Among them, the low-pass filter is used to filter the high-frequency part of the signal spectrum; the high-pass is used to filter the low-frequency part of the signal spectrum; the role of the low-pass filter and the high-pass filter is to filter the bands of different frequencies of the signal.

As shown in FIG3 , the audio signals of the left and right channels LR after the total volume adjustment are first obtained, and the audio signals of the LR are subjected to channel mixing processing. After that, the mixed audio signals are subjected to level detection, which can be detected by a level detection module, and the detected level value is transformed with a plurality of different coefficients set, that is, the level value is increased to different frequency bands. Finally, the increased signal is converted into a corresponding control signal to control the gain of the two signals led out by the amplifier to obtain a gain signal, and different signals are processed after passing through filters of different frequency bands, and the obtained signal is mixed with the original corresponding L or R channel, so as to output an audio signal after multi-band dynamic compensation. Digital signal processing not only saves costs, but also achieves the effect of multi-level compensation.

In FIG. 3 , the real-time performance and smoothness of signal detection can be changed by changing the response time and release time of level detection.

The coefficient transformation in FIG3 can change the value of the subsequent signal gain according to the detected level value, specifically including: when the level value is less than or equal to the set threshold, the gain takes effect, and the smaller the level value is, the greater the gain is. In addition, for the same input level, the gain control obtained by the coefficient transformation module in different frequency bands is also different. For example, the gain of the low-pass frequency band is generally larger than that of the high-pass frequency band, which is obtained by the equal loudness curve.

It should be noted that the multi-stage compensation part in FIG. 3 is a bandpass filter, which can be expanded into multiple bandpass filters so that users can compensate for different frequency bands according to their needs, thereby achieving a multi-stage compensation effect.

The specific implementation process of the level detection and coefficient transformation in FIG. 3 in the DSP is described in detail below. This embodiment is described by taking a code implementation as an example.

1) The code implementation process of level detection includes:

Among them, the level detection in this embodiment needs to be performed on a small signal, so it is more appropriate to take the envelope of the signal, for example, the envelope of its positive half cycle can be taken.

#defineFS48000

if(InputDetect>0&&InputDetect>HoldDetect)

{

IsRelease=0;

HoldDetect = InputDetect;

HoldTime=0;

That is to say, once the level value of the input audio signal is greater than the stored value, the level value is immediately updated and stored, the release state is stopped and the timing is restarted, otherwise, the timing continues.

}else HoldTime++;

if(HoldTime>ReleaseTime)

{

IsRelease=1;

In this program segment, the release time (ReleaseTime) is an adjustable parameter. Assuming that the sampling rate is 48KHZ at this time, when ReleaseTime is 48000, the system should maintain the level for 1 second.

}

if(IsRelease)

{

HoldDetect=HoldDetect*ReleaseRate;

That is to say, the release rate (ReleaseRate) is a decimal between 0 and 1. Once the release begins, since the input audio signal is smaller than the saved signal value, the signal will naturally decline. The ReleaseRate is the decline rate of each sampling point, and the unit is G/Ts.

}

OutputDetect = HoldDetect;

That is to say, when the release is not turned on, the saved value is not output as a drop, and when it is turned on, it is output after dropping.

2) The code implementation process of the controllable gain of coefficient transformation includes:

InputRadioChange = OutputDetect;

CurrentLevel= 1650*InputRadioChange/(2^23);

//Since the DSP currently uses a 24-bit sampling accuracy, there is one bit for the sign bit, and its center voltage is 1650mV. Therefore, the unit of CurrentLevel in the above formula is mV.

if(CurrentLevel>=ChangeLevel)

// ChangeLevel is an adjustable parameter in mV. Once the detected signal is less than this value, the dynamic gain will be

{

DyGainRadio=0;

// DyGainRadio is a dynamic factor between 0 and 1, which interacts with the subsequent maximum gain MaxGain to form a dynamic gain

}else

{

DyGainRadio = (ChangeLevel - CurrentLevel) / ChangeLevel;

//The smaller the detected value, the larger the dynamic factor

}

DyGain= DyGainRadio*MaxGain;

//In the above formula, MaxGain is the maximum adjustable gain. By multiplying it with the dynamic factor, we can get the dynamic gain DyGain that is ultimately used for volume adjustment.

In this embodiment, the level value after volume adjustment is detected based on the DSP, and the compensation curve of the low-intermediate frequency is determined by the level value after volume adjustment. The equal loudness is affected by controlling the input level value and the gain size inside the DSP, and then mixed with the original LR channel after passing through different filters, so as to achieve the effect of multi-band dynamic compensation.

For example, a certain car head unit has 43 volume levels. When the volume is below 40, equal loudness will work. At this time, whether you increase or decrease the volume of the input sound source, as long as the volume value of the head unit is not changed, equal loudness will work. It is wrong to still use equal loudness when increasing the volume of the input sound source. When decreasing the volume of the input sound source, since the sound pressure level from the speaker has been reduced, it is not enough to compensate according to the original equal loudness curve.

Based on this, in this embodiment, an improvement is made on the basis of the traditional equal loudness. The level value after volume adjustment is detected by the DSP to determine the compensation curve. At this time, the input level value and the gain size inside the DSP can jointly affect the equal loudness, which can solve the problems existing in the traditional equal loudness. The so-called equal loudness means that if two frequency bands are to be enhanced, the smaller the sound, the greater the enhancement.

Please also refer to Figure 4, which is a schematic diagram of another application example of a method for processing an audio signal provided by an embodiment of the present invention, wherein the method performs compensation processing on the audio signals of the left and right channels through virtual surround sound. This embodiment takes the acquisition of the audio signals of the left channel L and the right channel R as an example. As shown in Figure 4, it includes: an inverter, an adder, a spatial sound controller, a central sound controller, a frequency compensator, a left mixer and a right mixer, etc. Virtual surround sound, also called virtual stereo, stereo widening, virtual 3D sound field, SRS, is a technology that expands stereo into multi-channel.

In this embodiment, for example, to reproduce the 3D sound field, the ambient sound signal recorded in the ordinary stereo signal is first extracted, and then the spectrum of the extracted ambient sound signal is changed accordingly according to the frequency response function of the human ear. The processed signal is still a two-channel audio signal, and after it is sent to the power amplifier, the 3D sound field can be reproduced.

In this embodiment, the virtual surround sound is realized by the auditory space positioning mechanism. By analyzing the auditory space positioning mechanism, the following conclusions can be drawn:

(a) In the case of medium and low frequencies (less than about 1.5 kHz), the interaural time difference or phase difference is the main factor in positioning:

(b) In the case of medium frequencies (about 1.5 kHz to 4.0 kHz), the interaural time difference and intensity difference play a role in localization;

(c) In the case of mid-high frequencies (about 4 kHz to 5 kHz), the interaural intensity difference is the main factor in localization;

(d) At high frequencies (approximately greater than 5 kHz to 6 kHz), the pinna effect plays an important role in distinguishing the sound source at the front and rear mirror positions and the positioning of the median vertical plane;

(e) The changes in intensity and time differences between the ears caused by the listener's involuntary slight rotation of the head play an important role in distinguishing the sound sources in the front and rear mirror positions.

(f) In a rectifier environment, the sound intensity perceived by the human ear is the main factor in locating the distance of the sound source; in a reverberant environment, the ratio of the sound energy of the direct sound to that of the reverberant sound plays a major role in distance positioning.

It should be noted that if two microphones are placed on the left and right sides of the stage for recording left and right channels, when a musical instrument makes a sound at a certain point in the instrument area, unless the instrument is on the midline of the two microphones, the sound it emits will inevitably produce a time difference (phase difference) when entering the two microphones due to the different distances.

Based on this, this embodiment extracts the phase difference of this segment by subtracting the left and right channels of the sound source, and then performs frequency compensation based on the conclusion of the auditory spatial positioning mechanism (for example, in the case of mid-low frequencies (approximately less than 1.5KHz), the inter-aural time difference or phase difference is the main factor for positioning), thereby highlighting the mid-low frequencies, increasing its spatial positioning capability, and simulating the surround sound channel. By changing its level, it can highlight the ambient sound and various emitted sounds, and improve the sense of space and movement.

Similarly, you can also simulate the center channel by adding the left and right channels, and change its sound image positioning and front sound field. Among them, the center output generally refers to an output channel of the audio. Generally speaking, 2.1 channels only have left and right channels and bass output, and 6.1 channels have left and right channels (front), left and right channels (rear), center output, bass output, etc.

As shown in FIG4 , in this embodiment, on the one hand, the acquired audio signal of the right channel is added to the audio signal of the left channel through an inverter, and the phase difference is frequency compensated through the set spatial sound control to obtain the right channel audio signal after mid-low frequency compensation; and the audio signal of the right channel after frequency compensation is input into the right mixer through the inverter in one way, and directly input into the left mixer in the other way;

At the same time, on the other hand, the acquired left channel audio signal is directly added to the right channel audio signal to simulate the center audio signal, and the frequency of the center audio signal is controlled by the central sound control mechanism to obtain the left channel audio signal after sound image positioning, and the left channel audio signal is input into the amplifier for signal amplification, and the amplified voice signal is input into the right mixer in one way and the left mixer in the other way.

The right mixer performs right mixing processing on the original audio signal of the right channel, the audio signal input by the inverter, and the audio signal input by the left channel, thereby improving the spatial sense and movement sense of the surround sound, and the left mixer performs left mixing processing on the original audio signal of the left channel, the audio signal input by the amplifier, and the audio signal input by the right channel after frequency compensation, thereby improving the sound image positioning and the effect of the front sound field.

The specific implementation process of the frequency compensation in Figure 4 in DSP is described in detail below. This embodiment takes a frequency compensation implementation code as an example. The implementation of other modules in Figure 4 is relatively simple and is already a well-known technology for those skilled in the art.

Assume that the frequency compensation required by this embodiment is as shown in FIG. 5 , which is a schematic diagram of a frequency compensation provided by an embodiment of the present invention.

Based on the results of a large number of simulations in Matlab, it can be concluded that the three-band EQ can roughly simulate this curve, and its gain, center frequency, and Q value are {+10dB200Hz0.41; -2dB3000Hz0.41; +5dB20000Hz0.41} respectively. The specific implementation process includes:

Generally, to implement EQ, this embodiment uses the Peaking filter in the IIR filter as an example. The general form of the IIR filter is:

Its time domain expression is:

The parameters of the Peaking filter are

Among them,/>

Where G is the gain of the center frequency, in times; fc is the center frequency; fs is the sampling frequency; Q is the quality factor; a ₀ , a ₁ , a ₂ are the forward coefficients, b ₀ , b ₁ , b ₂ are the feedback coefficients; the intermediate variable is calculated using the center frequency fc and the sampling rate fs ,/> S,/> C, use Q value to calculate the variable/> , and use the gain value G to calculate the variable A.

Therefore, through the above formula, the various parameters of the three-band EQ can be calculated, and then the above time domain expression can be implemented by DSP to achieve frequency compensation.

It should be noted that the basic working principle of the IIR filter is to perform convolution operations with specified time domain coefficients, set frequency domain coefficients according to the filter characteristics, estimate the frequency domain characteristics of the original signal through the sampling point transfer function, and obtain the frequency domain formula of the signal to the output filter, and then transform the frequency domain result back to the time domain to achieve the filtering operation. This is a well-known technology for those skilled in the art and will not be repeated here.

The embodiment of the present invention utilizes digital signal processing technology, occupies less DSP resources, has low cost, good effect, high cost performance, and is also very simple to implement.

It should be noted that, for the method embodiments, for the sake of simplicity, they are all described as a series of action combinations, but those skilled in the art should know that this implementation disclosure is not limited by the order of the actions described, because according to the present invention, certain steps can be performed in other orders or simultaneously. Secondly, those skilled in the art should also know that the embodiments described in the specification are all preferred embodiments, and the actions involved are not necessarily required by the present invention.

FIG6 is a block diagram of an audio signal processing device provided by an embodiment of the present invention. Referring to FIG6 , the device includes: a first acquisition module 601, an adjustment module 602, a processing module 603, a superposition module 604 and an output module 605, wherein:

A first acquisition module 601 is used to acquire an audio signal after left and right channels are mixed;

An adjustment module 602, configured to adjust the level value of the audio signal based on digital signal processing;

A processing module 603 is used to perform dynamic bass enhancement processing on the audio signals corresponding to the left and right channels based on the adjusted level value;

A superposition module 604 is used to superimpose the processed audio signals of the left and right channels with the original audio signals of the corresponding channels respectively;

The output module 605 is used to output the left and right channel audio signals after superposition processing.

Optionally, the adjustment module includes:

A detection module, configured to detect a level value of a bass volume in the audio signal according to a response time and a release time of the level based on digital signal processing;

The level adjustment module is used to adjust the level value according to different coefficient transformations.

Optionally, the level adjustment module is specifically configured to multiply the level value by different set coefficients to adjust the level value of the bass volume in the audio signal.

Optionally, the processing module includes:

A gain adjustment module, used for adjusting the gain of the audio signal corresponding to the left and right channels based on the adjusted level value;

The compensation processing module is used to input the audio signals of the left and right channels after gain adjustment into corresponding filters for compensation processing of different frequency bands.

Optionally, the gain adjustment module is specifically configured to multiply the adjusted level value by a dynamic factor, and adjust the dynamic gain of volume adjustment in the audio signals corresponding to the left and right channels based on the multiplication result;

The compensation processing module is specifically used to input the audio signals of the left and right channels after adjusting the gains into corresponding low-pass filters and/or high-pass filters respectively for compensation processing in different frequency bands; or, input the audio signals of the left and right channels after adjusting the gains into corresponding one or more band-pass filters for multi-stage compensation processing in different frequency bands.

Optionally, the device further comprises:

The second acquisition module is used to acquire the audio signals of the left and right channels within a set time period after superposition processing;

An operation processing module, used to perform operation processing on the acquired audio signals of the left and right channels respectively, to obtain the phase difference and center channel signal of the corresponding channels after processing;

A virtual surround processing module, used to perform virtual surround sound processing on the processed phase difference and center channel signal respectively, to obtain a left channel audio signal after sound image localization and a right channel audio signal after mid-low frequency compensation;

The mixing processing module is used to perform left-right mixing processing on the right channel audio signal after mid-low frequency compensation, the left channel audio signal after sound image localization, and the corresponding left and right channel original audio signals, and output the left channel audio signal after left mixing processing and the right channel audio signal after right mixing processing.

Optionally, the operation processing includes:

A subtraction processing module, used to subtract the acquired audio signals of the left and right channels to obtain a phase difference of the audio signal within the set time period;

The addition processing module is used to add the acquired audio signals of the left and right channels to simulate the center channel signal within the set time period.

Optionally, the virtual surround processing module includes:

A frequency compensation processing module, used to perform frequency compensation on the phase difference according to the conclusion of the auditory space positioning mechanism, and obtain a right channel audio signal after mid-low frequency compensation;

The frequency control processing module is used to perform frequency control on the center audio signal through a central sound control mechanism to obtain a left channel audio signal after sound image localization.

Optionally, the frequency compensation processing module is specifically used to perform frequency compensation on the phase difference using a Peaking filter based on the conclusion of the auditory space localization mechanism to obtain a right channel audio signal after mid- and low-frequency compensation.

FIG7 is another block diagram of an audio signal processing device provided by an embodiment of the present invention. The device comprises: an acquisition module 701, an operation processing module 702, a virtual sound processing module 703, a mixing processing module 704 and an output module 705, wherein:

The acquisition module 701 is used to acquire the audio signals of the left and right channels within a set time period, wherein the audio signals include: the audio signals of the left and right channels before the left and right channels are mixed; or the audio signals of the left and right channels after the output superposition processing;

The operation processing module 702 is used to perform operation processing on the obtained audio signals of the left and right channels respectively to obtain the phase difference and center channel signal of the corresponding channels after processing;

A virtual sound processing module 703 is used to perform virtual surround sound processing on the processed phase difference and center channel signal to obtain a left channel audio signal after sound image localization and a right channel audio signal after mid-low frequency compensation;

A mixing processing module 704 is used to perform left-right mixing processing on the right channel audio signal after mid-low frequency compensation, the left channel audio signal after sound image localization, and the corresponding left and right channel original audio signals respectively;

The output module 705 is used to output the left channel audio signal after the left mixing process and the right channel audio signal after the right mixing process.

Optionally, the operation processing includes:

A subtraction processing module, used to subtract the acquired audio signals of the left and right channels to obtain a phase difference of the audio signal within the set time period;

The addition processing module is used to add the acquired audio signals of the left and right channels to simulate the center channel signal within the set time period.

Optionally, the virtual surround processing module includes:

A frequency compensation processing module, used to perform frequency compensation on the phase difference according to the conclusion of the auditory space positioning mechanism, and obtain a right channel audio signal after mid-low frequency compensation;

The frequency control processing module is used to perform frequency control on the center audio signal through a central sound control mechanism to obtain a left channel audio signal after sound image localization.

Optionally, the frequency compensation processing module is specifically used to perform frequency compensation on the phase difference using a Peaking filter based on the conclusion of the auditory space localization mechanism to obtain a right channel audio signal after mid- and low-frequency compensation.

Regarding the device in the above embodiment, the specific manner in which each module performs operations has been described in detail in the embodiment of the method, and will not be elaborated here.

The device embodiments described above are merely illustrative, wherein the units described as separate components may or may not be physically separated, and the components displayed as units may or may not be physical units, that is, they may be located in one place, or they may be distributed on multiple network units. Some or all of the modules may be selected according to actual needs to achieve the purpose of the scheme of this embodiment. Those of ordinary skill in the art may understand and implement it without creative work.

FIG8 is a block diagram of an electronic device 800 provided in an embodiment of the present invention. For example, the electronic device 800 may be a mobile terminal or a server, and the electronic device is a mobile terminal as an example for explanation in the embodiment of the present invention. For example, the electronic device 800 may be a mobile phone, a computer, a digital broadcast terminal, a message transceiver device, a game console, a tablet device, a medical device, a fitness device, a personal digital assistant, etc.

8 , electronic device 800 may include one or more of the following components: a processing component 802 , a memory 804 , a power component 806 , a multimedia component 808 , an audio component 810 , an input/output (I/O) interface 812 , a sensor component 814 , and a communication component 816 .

The processing component 802 generally controls the overall operation of the electronic device 800, such as operations associated with display, phone calls, data communications, camera operations, and recording operations. The processing component 802 may include one or more processors 820 to execute instructions to complete all or part of the steps of the above-mentioned method. In addition, the processing component 802 may include one or more modules to facilitate the interaction between the processing component 802 and other components. For example, the processing component 802 may include a multimedia module to facilitate the interaction between the multimedia component 808 and the processing component 802.

The memory 804 is configured to store various types of data to support operations on the device 800. Examples of such data include instructions for any application or method operating on the electronic device 800, contact data, phone book data, messages, pictures, videos, etc. The memory 804 can be implemented by any type of volatile or non-volatile storage device or a combination thereof, such as static random access memory (SRAM), electrically erasable programmable read-only memory (EEPROM), erasable programmable read-only memory (EPROM), programmable read-only memory (PROM), read-only memory (ROM), magnetic memory, flash memory, magnetic disk or optical disk.

The power supply component 806 provides power to the various components of the electronic device 800. The power supply component 806 may include a power management system, one or more power supplies, and other components associated with generating, managing, and distributing power to the electronic device 800.

The multimedia component 808 includes a screen that provides an output interface between the electronic device 800 and the user. In some embodiments, the screen may include a liquid crystal display (LCD) and a touch panel (TP). If the screen includes a touch panel, the screen may be implemented as a touch screen to receive input signals from the user. The touch panel includes one or more touch sensors to sense touch, slide, and gestures on the touch panel. The touch sensor may not only sense the boundaries of the touch or slide action, but also detect the duration and pressure associated with the touch or slide operation. In some embodiments, the multimedia component 808 includes a front camera and/or a rear camera. When the device 800 is in an operating mode, such as a shooting mode or a video mode, the front camera and/or the rear camera may receive external multimedia data. Each front camera and rear camera may be a fixed optical lens system or have a focal length and optical zoom capability.

The audio component 810 is configured to output and/or input audio signals. For example, the audio component 810 includes a microphone (MIC), and when the electronic device 800 is in an operation mode, such as a call mode, a recording mode, and a voice recognition mode, the microphone is configured to receive an external audio signal. The received audio signal can be further stored in the memory 804 or sent via the communication component 816. In some embodiments, the audio component 810 also includes a speaker for outputting audio signals.

I/O interface 812 provides an interface between processing component 802 and peripheral interface modules, such as keyboards, click wheels, buttons, etc. These buttons may include, but are not limited to, home buttons, volume buttons, start buttons, and lock buttons.

The sensor assembly 814 includes one or more sensors for providing various aspects of status assessment for the electronic device 800. For example, the sensor assembly 814 can detect the open/closed state of the device 800, the relative positioning of components, such as the display and keypad of the electronic device 800, and the sensor assembly 814 can also detect the position change of the electronic device 800 or a component of the electronic device 800, the presence or absence of user contact with the electronic device 800, the orientation or acceleration/deceleration of the electronic device 800, and the temperature change of the electronic device 800. The sensor assembly 814 may include a proximity sensor configured to detect the presence of nearby objects without any physical contact. The sensor assembly 814 may also include a light sensor, such as a CMOS or CCD image sensor, for use in imaging applications. In some embodiments, the sensor assembly 814 may also include an acceleration sensor, a gyroscope sensor, a magnetic sensor, a pressure sensor, or a temperature sensor.

The communication component 816 is configured to facilitate wired or wireless communication between the electronic device 800 and other devices. The electronic device 800 can access a wireless network based on a communication standard, such as WiFi, a carrier network (such as 2G, 3G, 4G or 5G), or a combination thereof. In an exemplary embodiment, the communication component 816 receives a broadcast signal or broadcast-related information from an external broadcast management system via a broadcast channel. In an exemplary embodiment, the communication component 816 also includes a near field communication (NFC) module to facilitate short-range communication. For example, the NFC module can be implemented based on radio frequency identification (RFID) technology, infrared data association (IrDA) technology, ultra-wideband (UWB) technology, Bluetooth (BT) technology and other technologies.

In an embodiment, the electronic device 800 may be implemented by one or more application-specific integrated circuits (ASICs), digital signal processors (DSPs), digital signal processing devices (DSPDs), programmable logic devices (PLDs), field programmable gate arrays (FPGAs), controllers, microcontrollers, microprocessors, or other electronic components to execute the audio signal processing method shown above.

In an embodiment, a computer-readable storage medium is also provided, and when the instructions in the computer-readable storage medium are executed by the processor of the electronic device, the electronic device 800 can perform the above-mentioned audio signal processing method. For example, the computer-readable storage medium can be a ROM, a random access memory (RAM), a CD-ROM, a magnetic tape, a floppy disk, an optical data storage device, etc.

In an embodiment, a computer program product is further provided, including a computer program or instructions. When the computer program or instructions are executed by the processor 820 of the electronic device 800, the electronic device 800 executes the above-mentioned audio signal processing method.

FIG9 is a block diagram of a device 900 for processing an audio signal provided by an embodiment of the present invention. For example, the device 900 can be provided as a server. Referring to FIG9 , the device 900 includes a processing component 922, which further includes one or more processors, and a memory resource represented by a memory 932, for storing instructions that can be executed by the processing component 922, such as an application. The application stored in the memory 932 can include one or more modules each corresponding to a set of instructions. In addition, the processing component 922 is configured to execute instructions to perform the above method.

The device 900 may also include a power supply component 926 configured to perform power management of the device 900, a wired or wireless network interface 950 configured to connect the device 900 to a network, and an input/output (I/O) interface 958. The device 900 may operate based on an operating system stored in the memory 932, such as Windows Server™, Mac OS X™, Unix™, Linux™, FreeBSD™, or the like.

FIG10 is a block diagram of an electronic device provided by an embodiment of the present invention. As shown in the figure, it includes a processor 1001, a communication interface 1002, a memory 1003 and a communication bus 1004, wherein the processor 1001, the communication interface 1002, and the memory 1003 communicate with each other through the communication bus 1004;

Memory 1003, used to store the processor executable instructions;

The processor 1001 is used to implement the above method when executing the executable instructions in the memory 1003.

The communication bus in this embodiment may be a Peripheral Component Interconnect (PCI) bus or an Extended Industry Standard Architecture (EISA) bus, etc. The communication bus may be divided into an address bus, a data bus, a control bus, etc. For ease of representation, only one thick line is used in the figure, but it does not mean that there is only one bus or one type of bus.

The communication interface is used for communication between the above electronic device and other devices.

The memory may include a random access memory (RAM) or a non-volatile memory, such as at least one disk memory. Optionally, the memory may also be at least one storage device located away from the aforementioned processor.

The above-mentioned processor can be a general-purpose processor, including a central processing unit (CPU), a network processor (NP), etc.; it can also be a digital signal processor (DSP), an application specific integrated circuit (ASIC), a field programmable gate array (FPGA) or other programmable logic devices, discrete gate or transistor logic devices, discrete hardware components.

In another embodiment of the present invention, a computer-readable storage medium is provided. When instructions in the computer-readable storage medium are executed by a processor of an electronic device, the electronic device is enabled to execute the method described above.

In another embodiment of the present invention, a computer program product including a computer program or instructions is provided. When the computer program or instructions are executed by a processor of an electronic device, the method for processing an audio signal as described above is implemented.

In an embodiment, a computer-readable storage medium is also provided, and when the instructions in the computer-readable storage medium are executed by a processor of an electronic device, the electronic device can perform the above-mentioned audio signal processing method. For example, the computer-readable storage medium can be a ROM, a random access memory (RAM), a CD-ROM, a magnetic tape, a floppy disk, an optical data storage device, etc.

In an embodiment, a computer program product is further provided, including a computer program or instructions. When the computer program or instructions are executed by the processor 1001 of the electronic device, the electronic device executes the above-mentioned method for processing audio signals.

Those skilled in the art will readily appreciate other embodiments of the present invention after considering the specification and practicing the invention disclosed herein. This application is intended to cover any variations, uses or adaptations of the present invention that follow the general principles of the present invention and include common knowledge or customary techniques in the art that are not disclosed by the present invention. The specification and examples are to be considered exemplary only, and the true scope and spirit of the present invention are indicated by the following claims.

It should be understood that the present invention is not limited to the exact construction that has been described above and shown in the drawings and that various modifications and changes may be made without departing from the scope thereof. The scope of the present invention is limited only by the appended claims.

Claims (14)

1. A method of processing an audio signal, comprising:

Acquiring an audio signal after mixing left and right channels;

Adjusting the magnitude of a level value in the audio signal based on digital signal processing;

based on the level value, performing dynamic bass enhancement processing on the audio signals corresponding to the left and right channels;

Respectively carrying out superposition processing on the processed audio signals of the left channel and the right channel and the original audio signals of the corresponding channels;

And outputting the audio signals of the left and right channels after the superposition processing.

2. The method according to claim 1, wherein the adjusting the magnitude of the level value in the audio signal based on the digital signal processing includes:

Detecting a level value of a low volume in the audio signal according to a response time and a release time of the level based on digital signal processing;

the level values are sized according to different coefficient transforms.

3. The method of processing an audio signal according to claim 2, wherein said adjusting the magnitude of the level value according to different coefficient transforms comprises:

the level value is multiplied by a set different coefficient to adjust the level value of the low volume in the audio signal.

4. A processing method of an audio signal according to any one of claims 1 to 3, wherein the performing dynamic bass enhancement processing on the audio signal corresponding to the left and right channels based on the magnitude of the adjusted level value includes:

adjusting the gain of the audio signal corresponding to the left and right channels based on the level value;

and inputting the audio signals of the left and right channels after gain adjustment into corresponding filters to perform compensation processing of different frequency bands.

5. The method according to claim 4, wherein adjusting the gain of the audio signal corresponding to the left and right channels based on the magnitude of the adjusted level value comprises: multiplying the level value by a dynamic factor based on the level value, and adjusting the dynamic gain of volume adjustment in the audio signals corresponding to the left and right channels based on the multiplication result;

The step of inputting the audio signals of the left and right channels after gain adjustment into corresponding filters to perform compensation processing of different frequency bands comprises the following steps: respectively inputting the audio signals of the left and right channels after gain adjustment to corresponding low-pass filters and/or high-pass filters to perform compensation processing of different frequency bands; or inputting the audio signals of the left and right channels after gain adjustment to one or more corresponding band-pass filters to carry out multi-stage compensation processing of different frequency bands.

6. The method of processing an audio signal according to claim 1, characterized in that the method further comprises:

Acquiring audio signals of left and right channels in a set time period after superposition processing;

Respectively carrying out operation processing on the acquired audio signals of the left channel and the right channel to obtain a phase difference and a middle channel signal of the corresponding channel after processing;

Respectively carrying out virtual surround sound processing on the processed phase difference and the processed center channel signal to obtain a left channel audio signal with positioned sound image and a right channel audio signal with middle-low frequency compensation;

and respectively performing left and right mixing processing on the right channel audio signal subjected to middle and low frequency compensation and the left channel audio signal subjected to sound image positioning and the corresponding left and right channel original audio signals, and outputting a left channel audio signal subjected to left mixing processing and a right channel audio signal subjected to right mixing processing.

7. The method for processing audio signals according to claim 6, wherein the performing an arithmetic process on the acquired audio signals of the left and right channels, respectively, to obtain a phase difference of the processed corresponding channels and a center channel signal, includes:

subtracting the acquired audio signals of the left and right channels to obtain a phase difference of the audio signals in the set time period;

And adding the acquired audio signals of the left channel and the right channel, and simulating a middle-set channel signal in the set time period.

8. The method for processing audio signals according to claim 6, wherein the performing virtual surround processing on the processed phase difference and center channel signals to obtain a left channel audio signal with a sound image positioned and a right channel audio signal with a middle-low frequency compensated respectively includes:

frequency compensation is carried out on the phase difference through the conclusion of the auditory space positioning mechanism, and a right channel audio signal after middle-low frequency compensation is obtained;

And performing frequency control on the middle-set audio signal through a central sound control mechanism to obtain a left channel audio signal with a sound image positioned.

9. The method for processing an audio signal according to claim 8, wherein the step of frequency compensating the phase difference by the conclusion of the auditory spatial localization mechanism to obtain a low-and-medium frequency compensated right channel audio signal comprises:

And carrying out frequency compensation on the phase difference by using Peaking filters according to the conclusion of the auditory space positioning mechanism to obtain a right channel audio signal after medium-low frequency compensation.

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

Acquiring audio signals of left and right channels within a set time period, wherein the audio signals comprise: audio signals of left and right channels before mixing of the left and right channels; or outputting the audio signals of the left and right channels after superposition processing;

Respectively processing the acquired audio signals of the left channel and the right channel to obtain a phase difference and a middle channel signal of the corresponding channels after processing;

Respectively carrying out virtual surround sound processing on the processed phase difference and the processed center channel signal to obtain a left channel audio signal with positioned sound image and a right channel audio signal with middle-low frequency compensation;

Respectively performing left and right mixing processing on the right channel audio signal subjected to middle and low frequency compensation, the left channel audio signal subjected to sound image positioning and the corresponding left and right channel original audio signals;

And outputting the left channel audio signal after left mixing processing and the right channel audio signal after right mixing processing.

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

the first acquisition module is used for acquiring the audio signal after the left and right channels are mixed;

An adjusting module for adjusting the level value in the audio signal based on digital signal processing;

The processing module is used for carrying out dynamic bass enhancement processing on the audio signals corresponding to the left and right channels based on the adjusted level value;

the superposition module is used for respectively superposing the processed audio signals of the left channel and the right channel with the original audio signals of the corresponding channels;

And the output module is used for outputting the audio signals of the left and right channels after the superposition processing.

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

The acquisition module is used for acquiring audio signals of left and right channels within a set time period, and the audio signals comprise: audio signals of left and right channels before mixing of the left and right channels; or outputting the audio signals of the left and right channels after superposition processing;

The operation processing module is used for respectively performing operation processing on the acquired audio signals of the left channel and the right channel to obtain a phase difference of the corresponding channel after processing and a centrally-mounted channel signal;

The virtual sound processing module is used for respectively carrying out virtual surround sound processing on the processed phase difference and the processed middle-set channel signal to obtain a left channel audio signal after sound image positioning and a right channel audio signal after middle-low frequency compensation;

the audio mixing processing module is used for respectively carrying out left and right audio mixing processing on the right channel audio signal subjected to the middle and low frequency compensation, the left channel audio signal subjected to sound image positioning and the corresponding left and right channel original audio signals;

And the output module is used for outputting the left channel audio signal after the left mixing processing and the right channel audio signal after the right mixing processing.

13. An electronic device, comprising:

A processor;

a memory for storing the processor-executable instructions;

wherein the processor is configured to execute the instructions to implement the method of processing an audio signal as claimed in any one of claims 1 to 10.

14. A computer readable storage medium, characterized in that instructions in the computer readable storage medium, when executed by a processor of an electronic device, enable the electronic device to perform the method of processing an audio signal according to any one of claims 1 to 10.