Audio signal processing method and device, electronic equipment and readable storage medium
By using N related bandpass filter banks for audio signal processing, the problem of low audio signal quality in existing technologies is solved, and higher quality audio signal output is achieved.
Patent Information
- Authority / Receiving Office
- CN · China
- Patent Type
- Patents(China)
- Current Assignee / Owner
- VIVO MOBILE COMM CO LTD
- Filing Date
- 2022-09-23
- Publication Date
- 2026-07-07
AI Technical Summary
The filters used in existing audio signal processing result in low audio signal quality and distortion problems.
A target bandpass filter bank, including N related first bandpass filters, is used to process audio signals through a linear phase FIR bandpass filter bank or a nonlinear phase IIR bandpass filter bank. The first bandpass filters are designed to be of even order or the filters are designed according to their order.
It improves the quality of audio signals, reduces sound quality loss during processing, and obtains higher quality audio signals.
Smart Images
Figure CN115547350B_ABST
Abstract
Description
Technical Field
[0001] This application belongs to the field of audio technology, specifically relating to an audio signal processing method, apparatus, electronic device, and readable storage medium. Background Technology
[0002] Currently, during the acquisition of audio signals such as calls, recordings, and videos, audio processing can be performed on the acquired audio signals to improve their signal quality. Among these processes, Dynamic Range Control (DRC) is widely used in audio signal processing and is a signal amplitude adjustment method that can make the sound softer or louder.
[0003] In related technologies, the acquired audio signal can be divided into several sub-bands, and dynamic range control can be applied to each sub-band separately. This is known as Multi-band Dynamic Range Control (MBDRC) technology. Specifically, the MBDRC process for acquiring the audio signal is as follows: First, the acquired audio signal is input into different bandpass filters to obtain sub-band signals of different frequencies. Then, DRC technology is used to adjust the amplitude of each sub-band signal. Finally, the processed sub-band signals are combined to obtain the processed audio signal.
[0004] However, the audio signal obtained after processing by the filters used in the above audio signal processing is of low quality and is distorted. Summary of the Invention
[0005] The purpose of this application is to provide an audio signal processing method, apparatus, electronic device, and readable storage medium, which can solve the problem that the processed audio signal is of low quality and has distortion after the filter is used to process the audio signal during the audio signal processing process.
[0006] In a first aspect, embodiments of this application provide an audio signal processing method, which includes: acquiring a first audio signal; inputting the first audio signal into a target bandpass filter bank for filtering to obtain a second audio signal, wherein the target bandpass filter bank includes N first bandpass filters, and the N first bandpass filters are correlated with each other; performing signal processing on the second audio signal to obtain a third audio signal; wherein, when the target bandpass filter bank is a linear phase FIR bandpass filter bank, the first bandpass filters are even-order filters; when the target bandpass filter bank is a nonlinear phase IIR bandpass filter bank, the first bandpass filters are designed according to their order.
[0007] Secondly, embodiments of this application provide an audio signal processing apparatus. The apparatus includes an acquisition module and a processing module. The acquisition module is used to acquire a first audio signal. The processing module is used to input the first audio signal into a target bandpass filter bank for filtering to obtain a second audio signal. The target bandpass filter bank includes N first bandpass filters, and the N first bandpass filters are correlated. The processing module is further used to perform signal processing on the second audio signal to obtain a third audio signal. Wherein, when the target bandpass filter bank is a linear-phase FIR bandpass filter bank, the first bandpass filters are even-order filters. The processing module is further used to design a first bandpass filter based on the order of the first bandpass filter when the target bandpass filter bank is a nonlinear-phase IIR bandpass filter bank.
[0008] Thirdly, embodiments of this application provide an electronic device including a processor and a memory, wherein the memory stores programs or instructions executable on the processor, and the programs or instructions, when executed by the processor, implement the steps of the method described in the first aspect.
[0009] Fourthly, embodiments of this application provide a readable storage medium on which a program or instructions are stored, which, when executed by a processor, implement the steps of the method described in the first aspect.
[0010] Fifthly, embodiments of this application provide a chip, the chip including a processor and a communication interface, the communication interface being coupled to the processor, the processor being used to run programs or instructions to implement the method as described in the first aspect.
[0011] In a sixth aspect, embodiments of this application provide a computer program product stored in a storage medium, which is executed by at least one processor to implement the method described in the first aspect.
[0012] In this embodiment, a first audio signal is acquired; the first audio signal is then input into a target bandpass filter group for filtering to obtain a second audio signal. The target bandpass filter group includes N first bandpass filters, which are correlated with each other. The second audio signal is then processed to obtain a third audio signal. Wherein, if the target bandpass filter group is a linear-phase FIR bandpass filter group, the first bandpass filters are even-order filters; if the target bandpass filter group is a nonlinear-phase IIR bandpass filter group, the first bandpass filters are designed according to their order. Thus, since the N groups of first bandpass filters in the target bandpass filter group are correlated rather than independent, a higher quality audio signal is obtained, effectively reducing the loss of sound quality during audio signal processing. Attached Figure Description
[0013] Figure 1 This is a schematic flowchart of an audio signal processing method provided in an embodiment of this application;
[0014] Figure 2 This is a schematic diagram illustrating an example of a prior art audio signal processing method provided in this application embodiment;
[0015] Figure 3 This is one of the schematic diagrams illustrating the design flow of a bandpass filter bank in an audio signal processing method provided in this application embodiment;
[0016] Figure 4 This is the second schematic diagram of the design flow of a bandpass filter bank in an audio signal processing method provided in this application embodiment;
[0017] Figure 5 This is one of the example diagrams of an audio signal processing method provided in the embodiments of this application;
[0018] Figure 6 This is a second example diagram of an audio signal processing method provided in an embodiment of this application;
[0019] Figure 7 This is the third example diagram of an audio signal processing method provided in the embodiments of this application;
[0020] Figure 8 This is a schematic diagram of the structure of an audio signal processing device provided in an embodiment of this application;
[0021] Figure 9 This is one of the hardware structure diagrams of an electronic device provided in the embodiments of this application;
[0022] Figure 10 This is a second schematic diagram of the hardware structure of an electronic device provided in an embodiment of this application. Detailed Implementation
[0023] The technical solutions of the embodiments of this application will be clearly described below with reference to the accompanying drawings. Obviously, the described embodiments are only some, not all, of the embodiments of this application. All other embodiments obtained by those skilled in the art based on the embodiments of this application are within the scope of protection of this application.
[0024] The terms "first," "second," etc., used in the specification and claims of this application are used to distinguish similar objects and not to describe a specific order or sequence. It should be understood that such use of data can be interchanged where appropriate so that embodiments of this application can be implemented in orders other than those illustrated or described herein, and the objects distinguished by "first," "second," etc., are generally of the same class and the number of objects is not limited; for example, a first object can be one or more. Furthermore, in the specification and claims, "and / or" indicates at least one of the connected objects, and the character " / " generally indicates that the preceding and following objects are in an "or" relationship.
[0025] The audio signal processing method, apparatus, electronic device, and readable storage medium provided in this application will be described in detail below with reference to the accompanying drawings and through specific embodiments and application scenarios.
[0026] Currently, in the process of acquiring audio signals such as calls, recordings, and videos, audio processing can be performed on the acquired audio signals to improve their signal quality. Among these processes, DRC (Digital Ratio Control) is widely used in audio signal processing and is a signal amplitude adjustment method that can make the sound sound softer or louder.
[0027] In related technologies, the acquired audio signal can be divided into several sub-bands, and dynamic range control can be applied to each sub-band separately. This can be achieved using MBDRC technology to process the acquired audio signal. Specifically, for example... Figure 2 As shown, the process of using MBDRC technology to process the acquired audio signal is as follows: First, the acquired audio signal is input into different bandpass filters to obtain sub-band signals of different frequencies. Then, DRC technology is used to adjust the amplitude of each sub-band signal. Finally, the processed sub-band signals are combined to obtain the processed audio signal.
[0028] However, the N bandpass filters used in the aforementioned audio signal processing are N independent Infinite Impulse Response (IIR) bandpass filters. While this offers advantages in complexity, the MBDRC only considers its own center frequency and bandwidth, without any correlation, and fails to take into account the perfect reconstruction characteristics of the filter bank design. Therefore, the processed audio signal quality is low and prone to distortion. Furthermore, the IIR bandpass filters in the MBDRC lack linear phase characteristics, and the frequency selectivity of low-order filters is also poor.
[0029] In this embodiment, a first audio signal is acquired; and the first audio signal is input into a target bandpass filter group for filtering to obtain a second audio signal. The target bandpass filter group includes N first bandpass filters, which are correlated with each other. The second audio signal is then processed to obtain a third audio signal. Wherein, if the target bandpass filter group is a linear-phase FIR bandpass filter group, the first bandpass filters are even-order filters; if the target bandpass filter group is a nonlinear-phase IIR bandpass filter group, the first bandpass filters are designed according to their order. Thus, since the N groups of first bandpass filters in the target bandpass filter group in this embodiment are correlated rather than independent, a higher quality audio signal is obtained, effectively reducing the loss of sound quality during audio signal processing.
[0030] The entity executing the audio signal processing method provided in this application embodiment can be an audio signal processing device, which can be an electronic device or a functional module in the electronic device.
[0031] The audio signal processing method provided in this application will be described below using an audio signal processing device as an example.
[0032] This application provides an audio signal processing method. Figure 1 A flowchart of an audio signal processing method provided in an embodiment of this application is shown. Figure 1 As shown, the audio signal processing method provided in this application embodiment may include the following steps 201 to 203.
[0033] Step 201: Obtain the first audio signal.
[0034] In the embodiments of this application, the first audio signal may be acquired when the electronic device uses the recording function, or it may be acquired in real time during a call on the electronic device, or it may be acquired when the electronic device uses the video recording function. This application does not impose any restrictions.
[0035] Step 202: Input the first audio signal into the target bandpass filter bank for filtering to obtain the second audio signal.
[0036] In this embodiment of the application, the target bandpass filter bank includes N first bandpass filters, where N is a positive integer.
[0037] In this embodiment of the application, the N first bandpass filters included in the target bandpass filter are correlated with each other.
[0038] In the embodiments of this application, the aforementioned N first bandpass filters can be linear phase bandpass filters or nonlinear phase bandpass filters.
[0039] For example, the linear phase bandpass filter described above can be a linear phase finite impulse response (FIR) bandpass filter.
[0040] For example, the aforementioned nonlinear phase bandpass filter can be a nonlinear phase IIR bandpass filter.
[0041] In the embodiments of this application, the above-mentioned N first bandpass filter groups satisfy the perfect reconstruction condition.
[0042] For example, the above N first bandpass filter banks satisfy the first formula.
[0043] For example, the first formula above is:
[0044] Where N is the number of bandpass filter banks;
[0045] H n (z), n = 1, 2, ..., N are the filter frequency responses of the corresponding subbands;
[0046] It is an all-pass filter.
[0047] For example, under an FIR design, It is a pure time-delay filter; in IIR design, It is usually a non-pure time delay filter.
[0048] Step 203: Perform signal processing on the second audio signal to obtain the third audio signal.
[0049] In this embodiment of the application, the audio signal processing device performs DRC signal processing on the second audio signal to obtain a third audio signal.
[0050] In one possible embodiment, when the target bandpass filter bank is a linear-phase FIR bandpass filter bank, the first bandpass filter is an even-order filter.
[0051] In another possible embodiment, when the target bandpass filter bank is a nonlinear phase IIR bandpass filter bank, the first bandpass filter is designed according to the order of the first bandpass filter.
[0052] In the audio signal processing method provided in this application embodiment, a first audio signal is acquired; the first audio signal is then input into a target bandpass filter group for filtering to obtain a second audio signal. The target bandpass filter group includes N first bandpass filters, which are correlated with each other. The second audio signal is then processed to obtain a third audio signal. Wherein, if the target bandpass filter group is a linear-phase FIR bandpass filter group, the first bandpass filters are even-order filters; if the target bandpass filter group is a nonlinear-phase IIR bandpass filter group, the first bandpass filters are designed according to their order. Thus, since the N groups of first bandpass filters in the target bandpass filter group in this application embodiment are correlated rather than independent, a higher quality audio signal is obtained, effectively reducing the loss of sound quality during audio signal processing.
[0053] Optionally, in this embodiment of the application, the above-mentioned step 202, "inputting the first audio signal into the target bandpass filter bank for filtering to obtain the second audio signal," includes the following step 202a:
[0054] Step 202a: According to N preset sub-bands, the first audio signal obtained above is input into the target bandpass filter bank for filtering processing to obtain the second audio signal.
[0055] Where N is a positive integer.
[0056] For example, the second audio signal includes N sub-band signals corresponding to the N preset sub-bands, wherein one preset sub-band corresponds to one first bandpass filter.
[0057] For example, the aforementioned N preset sub-bands may be determined based on M segmented frequencies within the frequency band corresponding to the first audio signal.
[0058] For example, the aforementioned preset sub-band can be a non-uniform division of the entire operating frequency band of the first audio signal, or it can be a uniform division.
[0059] For example, each of the above N preset sub-bands corresponds to a sub-band signal.
[0060] Optionally, in this embodiment of the application, before step 202a, "the first audio signal obtained above is input into the target bandpass filter bank for filtering according to N preset sub-bands to obtain the second audio signal", the audio signal processing method provided in this embodiment of the application further includes step 301:
[0061] Step 301: Determine the M segmented frequencies as passband cutoff frequencies and design M filter triples.
[0062] For example, one segmentation frequency corresponds to one filter triplet.
[0063] For example, the above filter triplet includes: a low-pass filter, a high-pass filter, and an all-pass filter.
[0064] For example, the above N preset subbands correspond to M segmentation frequencies. Where M = N-1, in other words, the number of segmentation frequencies is one less than the number of preset subbands.
[0065] For example, the first bandpass filter mentioned above includes at least one target filter, wherein one target filter corresponds to one filter in a filter triplet.
[0066] For example, when the at least one target filter is at least two target filters, the at least two filter triplets corresponding to the at least two target filters are different.
[0067] For example, a filter triplet (H) can be designed for each preset subband. mL (z),H mH (z),H mAP (z)), m=1,2,...,N. Where H mL (z) is a low-pass filter, H mH (z) is a high-pass filter, H mAP (z)=H mL (z)+H mH (z) is an all-pass filter, and m is the m-th preset subband. That is, design N filter triples for N preset subbands.
[0068] The following describes the target bandpass filter bank for designing the audio signal processing method provided in this application using five possible embodiments as examples.
[0069] In some possible implementations:
[0070] In this embodiment, the target bandpass filter bank is a linear-phase FIR bandpass filter bank.
[0071] Optionally, in this embodiment of the application, the audio signal processing method provided in this embodiment of the application further includes the following steps 401 to 403:
[0072] Step 401: Design the even-order first low-pass filter corresponding to the first segmentation frequency based on the target design method.
[0073] For example, the above M segmentation frequencies include a first segmentation frequency, and the above M filter triples include a first filter triplet.
[0074] For example, the first filter triplet includes: a first low-pass filter, a first full-pass filter, and a first high-pass filter.
[0075] For example, the above target design method can be based on the fir1 function in Matlab, and this application does not impose any restrictions.
[0076] For example, based on the order of the even-order first low-pass filter corresponding to the first segmentation frequency, the group delay of the first low-pass filter of the first filter triplet corresponding to the first segmentation frequency is calculated using the second formula.
[0077] For example, the second formula above is D m =L m / 2.
[0078] Among them, D m The group delay of the first low-pass filter;
[0079] L m This is the order of the first low-pass filter.
[0080] Step 402: Based on the group delay of the first low-pass filter, design the first all-pass filter.
[0081] For example, the first all-pass filter described above is
[0082] Step 403: Based on the first low-pass filter and the first full-pass filter described above, design the first high-pass filter.
[0083] For example, the above high-pass filter is H mH (z)=H mAP (z)-H mL (z).
[0084] For example, firstly, design a low-pass filter H. mL (z), m = 1, 2, ..., N, specifically, according to the order L of the low-pass filter m (The order is even) Design, then use formula D m =L m / 2 Calculate the group delay of the low-pass filter, and obtain the corresponding all-pass filter based on the group delay. Finally, the high-pass filter H is obtained. mH (z)=H mAP (z)-H mL (z). For example, assuming N is 4, the design process for the corresponding 4 bandpass filter banks can be referred to... Figure 3 The design process shown is used to achieve this.
[0085] It should be noted that the filter order corresponding to the above N preset subbands is directly proportional to the subband length. In other words, the longer the subband length, the higher the filter order, meaning the steeper the frequency selectivity of the subband, and the higher the corresponding filter order. Conversely, the shorter the subband length, the lower the filter order.
[0086] Optionally, in this embodiment of the application, when the target bandpass filter bank is a linear-phase FIR bandpass filter bank, the above-mentioned step 202, "inputting the first audio signal into the target bandpass filter bank for filtering to obtain the second audio signal," includes the following steps 202b and 202c:
[0087] Step 202b: Concatenate the sub-filters in the target bandpass filter bank.
[0088] Step 202c: Input the first audio signal into the target bandpass filter bank after serial processing and filter it to obtain the second audio signal.
[0089] For example, the first audio signal is input into the target bandpass filter group, and DRC signal processing is performed on each sub-band signal in the preset sub-band corresponding to the sub-filter in the target bandpass filter group. Then, the processed N sub-band signals are combined to obtain the processed second audio signal.
[0090] In some possible implementations:
[0091] In this embodiment of the application, the target bandpass filter bank is a nonlinear phase IIR bandpass filter bank, the M filter triples include a second filter triplet, and the filter order of the second filter triplet is odd.
[0092] Optionally, in this embodiment of the application, the audio signal processing method provided in this embodiment of the application further includes the following steps 501 to 504:
[0093] Step 501: Design the first auxiliary filter bank based on the Butterworth filter design method.
[0094] For example, the second filter triplet mentioned above includes: a second low-pass filter, a second full-pass filter, and a second high-pass filter.
[0095] For example, the Butterworth filter design described above can be achieved using the butter function in Matlab.
[0096] For example, the first auxiliary filter bank mentioned above includes a first auxiliary low-pass filter and a first auxiliary high-pass filter.
[0097] Step 502: Based on the first auxiliary low-pass filter described above, design the second low-pass filter.
[0098] For example, when the filter order of the second filter triplet is odd, since the order of the first auxiliary low-pass filter is the same as the order of the second filter triplet, the second low-pass filter can be designed based on the order of the first auxiliary low-pass filter.
[0099] Step 503: Based on the first auxiliary high-pass filter described above, design the second high-pass filter.
[0100] For example, when the filter order of the second filter triplet is odd, the second high-pass filter is set based on the third formula and the first auxiliary high-pass filter.
[0101] For example, the third formula is H mH (z)=±H′ mH (z), H mL (z)=H′ mL (z).
[0102] Step 504: Based on the above-mentioned second low-pass filter and second high-pass filter, design the second all-pass filter.
[0103] For example, after setting the second low-pass filter and the second high-pass filter, the full-pass filter can be determined based on the low-pass filter and the high-pass filter. For example, H mAP (z)=H mL (z)+H mH (z).
[0104] It should be noted that the second filter triplet is an odd-order filter, and the order of the second filter triplet is the same as that of the first auxiliary filter group. The second low-pass filter is the aforementioned first auxiliary low-pass filter, and the aforementioned second high-pass filter is the aforementioned first auxiliary high-pass filter or the negative of the aforementioned first auxiliary high-pass filter.
[0105] In some possible implementations:
[0106] In this embodiment, the target bandpass filter bank is a nonlinear phase IIR bandpass filter bank, the M filter triplets include a third filter triplet, and the filter order of the third filter triplet is even.
[0107] Optionally, in this embodiment of the application, the audio signal processing method provided in this embodiment of the application further includes the following steps 601 to 604:
[0108] Step 601: Design the second auxiliary filter bank based on the Butterworth filter design method.
[0109] For example, the second auxiliary filter bank has an even order, and the filter order of the third filter triplet is twice the filter order of the second auxiliary filter bank.
[0110] For example, the second auxiliary filter bank mentioned above includes a second auxiliary low-pass filter and a second auxiliary high-pass filter.
[0111] Step 602: Based on the second auxiliary low-pass filter described above, design the third low-pass filter.
[0112] Step 603: Based on the second auxiliary high-pass filter described above, design the third high-pass filter.
[0113] For example, based on the orders of the second auxiliary low-pass filter and the second auxiliary high-pass filter described above, a third low-pass filter and a third high-pass filter are designed in conjunction with the fourth formula.
[0114] For example, the fourth formula is H mL (z)=H′ mL (z) 2 H mH (z)=H′ m H(z) 2 .
[0115] Step 604: Based on the above-mentioned third low-pass filter and third high-pass filter, design the third all-pass filter.
[0116] For example, after setting the third low-pass filter and the third high-pass filter, the all-pass filter can be determined based on these low-pass and high-pass filters. For example, H mAP (z)=H mL (z)+H mH (z).
[0117] For example, the aforementioned third filter triplet includes: a third low-pass filter, a third full-pass filter, and a third high-pass filter.
[0118] In one example, the third filter triplet is an even-order filter, the second auxiliary filter group is an even-order filter, the order of the third filter triplet is twice the order of the second auxiliary filter group, the third low-pass filter is the square of the second auxiliary low-pass filter, and the third high-pass filter is the square of the second auxiliary high-pass filter.
[0119] In some possible implementations:
[0120] In this embodiment, the target bandpass filter bank is a nonlinear phase IIR bandpass filter bank, the M filter triplets include a fourth filter triplet, and the filter order of the fourth filter triplet is even.
[0121] Optionally, in this embodiment of the application, the audio signal processing method provided in this embodiment of the application further includes the following steps 701 to 704:
[0122] Step 701: Design the third auxiliary filter bank based on the Butterworth filter design method.
[0123] For example, the third auxiliary filter bank has an odd order, and the filter order of the fourth filter triplet is twice that of the third auxiliary filter bank.
[0124] For example, the third auxiliary filter bank includes a third auxiliary low-pass filter and a third auxiliary high-pass filter.
[0125] Step 702: Based on the third auxiliary low-pass filter described above, design the fourth low-pass filter.
[0126] Step 703: Based on the third auxiliary high-pass filter described above, design the fourth high-pass filter.
[0127] For example, based on the orders of the third auxiliary low-pass filter and the third auxiliary high-pass filter mentioned above, the fourth low-pass filter and the fourth high-pass filter are designed in conjunction with the fifth formula.
[0128] For example, the fifth formula is H mL (z)=H′ mL (z) 2 H mH (z)=-H′ mH (z) 2 .
[0129] Step 704: Based on the above-mentioned fourth low-pass filter and fourth high-pass filter, design the fourth all-pass filter.
[0130] For example, after setting the fourth low-pass filter and the fourth high-pass filter, the all-pass filter can be determined based on these low-pass and high-pass filters. For example, H mAP (z)=H mL (z)+H mH (z).
[0131] For example, the fourth filter triplet mentioned above includes: a fourth low-pass filter, a fourth full-pass filter, and a fourth high-pass filter.
[0132] In one example, the fourth filter triplet is an even-order filter, the third auxiliary filter group is an odd-order filter, the order of the fourth filter triplet is twice the order of the third auxiliary filter group, the fourth low-pass filter is the square of the third auxiliary low-pass filter, and the third high-pass filter is the negative square of the third auxiliary high-pass filter.
[0133] For example, designing auxiliary nonlinear phase IIR low-pass and high-pass filters (H′) mL (z),H′ mH (z)), m=1,2,...,M, using the Butterworth filter design method (such as using the butter function in Matlab), the order can be odd or even. Since Butterworth low-pass and high-pass filters with the same cutoff frequency have complementary power characteristics, i.e., when the initial filter order is L... m At that time, if Then |A′ mL (ω)| 2 +|A′ mH (ω)| 2 =1. Therefore, theoretical analysis shows that when L′ m When H′ is even (i.e., the filter order of the second auxiliary filter bank mentioned above), mL (z) 2 +H′ mH (z) 2 It is an all-pass filter, therefore H can be chosen. mL (z)=H′ mL (z) 2 H mH (z)=H′ mH (z) 2 Each constitutes an even-order L m =2L′ m (i.e., the filter order of the third filter triplet mentioned above) filter; when L′ m When H′ is an odd number (i.e., the filter order of the third auxiliary filter bank mentioned above), mL (z) 2 -H′ mH (z)2 It is an all-pass filter, therefore H can be chosen. mL (z)=H′ mL (z) 2 H mH (z)=-H′ mH (z) 2 Each constitutes an even-order L m =2L′ m The filter (i.e., the filter order of the fourth filter triplet mentioned above); simultaneously, L′ m When H′ is an odd number (i.e., the filter order of the first auxiliary filter bank mentioned above), mL (z)±H′ mH (z) is an all-pass filter, therefore H can be chosen. mL (z)=H′ mL (z), H mH (z)=±H′ mH (z) respectively form odd-order L m =L′ m The filter (i.e., the filter order of the second filter triplet mentioned above). For example, assuming N is 4, the design process for the corresponding 4 bandpass filter banks can be found in [reference needed]. Figure 3 The design process shown is used to achieve this.
[0134] In some possible implementations:
[0135] In this embodiment of the application, the target filter is the high-pass filter in the target filter triplet.
[0136] Optionally, in this embodiment of the application, the audio signal processing method provided in this embodiment of the application further includes the following step 801:
[0137] Step 801: Replace the target filter with the all-pass filter in the target filter triplet.
[0138] In other words, reference Figure 4 In this embodiment, the target filter includes only a low-pass filter and a full-pass filter, thereby reducing the complexity of bandpass filter bank design.
[0139] In this way, N interconnected first bandpass filters based on preset subbands can be designed to divide the audio signal into N subband signals, and DRC signal processing can be performed on each of the N subband signals, thereby resulting in higher audio quality.
[0140] The following example illustrates the concept using an audio signal operating within a 24kHz frequency band. The MBDRC sub-band number N is set to 5, and the split frequencies are: 1kHz, 2kHz, 5kHz, and 10kHz.
[0141] In one possible implementation, it is implemented as an FIR type, with a total order L ≤ 1024 - "frame shift length", so that it can be implemented using a fast algorithm of 1024-point FFT. Figure 5 The design results for an FIR filter. Based on... Figure 5 Example results: the equivalent filter (i.e., the target filter mentioned above) H for each subband in the FIR filter bank. n (z), n=1,2,...,N has a relatively steep frequency selectivity. If the π flip of the phase spectrum is ignored, all phase spectrum curves are completely coincident and have the same constant group delay. The superposition of all subband filters “sum” is a pure time delay filter, thus meeting our lossless design requirements.
[0142] In one possible implementation, it is implemented as an IIR type, with M low-pass filters of order L. m =3, m=1,2,...,M and Figure 6 odd order L m Design results of IIR filter bank with a value of 3; Figure 7 Even order L m Design results of IIR filter bank with 4 = 4.
[0143] like Figure 6 and Figure 7 As shown, compared to FIR designs, the frequency selectivity of the equivalent filter in each subband of an IIR filter deteriorates, and it also loses its linear phase characteristic. However, its advantage is that it can be implemented with lower complexity than the FIR type, especially... Figure 4 The indirect implementation structure further reduces complexity. Furthermore, it can be seen that in adjacent-order IIR filter bank implementations, odd-order filters exhibit better frequency selectivity than even-order filters, but their phase spectrum consistency is worse. Both of these aspects fulfill our lossless filter bank design requirements (ignoring phase spectrum variations).
[0144] Thus, it can be seen that in terms of the non-uniform multi-band decomposition and synthesis (lossless) processing effect of audio, "the FIR filter bank of this application embodiment" > "the IIR filter bank of this application embodiment" > "independently designed several IIR bandpass filters"; while in terms of complexity, it is usually the opposite order. Therefore, in practical applications, there is a trade-off between high fidelity and complexity.
[0145] It should be noted that both FIR and IIR design processes support the setting of mixed odd and even orders.
[0146] It should be noted that the audio signal processing method provided in this application embodiment can be executed by an audio signal processing device, an electronic device, or a functional module or entity within an electronic device. This application embodiment uses an audio signal processing device executing the audio signal processing method as an example to illustrate the audio signal processing device provided in this application embodiment.
[0147] Figure 8 A schematic diagram of a possible structure of the audio signal processing apparatus involved in an embodiment of this application is shown. For example... Figure 8 As shown, the audio signal processing device 900 may include: an acquisition module 901 and a processing module 902; the acquisition module 901 is used to acquire a first audio signal; the processing module 902 is used to input the first audio signal into a target bandpass filter bank for filtering processing to obtain a second audio signal, the target bandpass filter bank including N first bandpass filters, and the N first bandpass filters are correlated with each other; the processing module 902 is also used to perform signal processing on the second audio signal to obtain a third audio signal; wherein, when the target bandpass filter bank is a linear phase FIR bandpass filter bank, the first bandpass filter is an even-order filter; the processing module 902 is also used to design a first bandpass filter according to the order of the first bandpass filter when the target bandpass filter bank is a nonlinear phase IIR bandpass filter bank.
[0148] Optionally, in this embodiment of the application, the processing module 902 is specifically used to input the first audio signal into the target bandpass filter bank for filtering according to N preset sub-bands to obtain the second audio signal; wherein, the second audio signal includes N sub-band signals corresponding to the N preset sub-bands, and one preset sub-band corresponds to one first bandpass filter.
[0149] Optionally, in this embodiment, the processing module 902 is further configured to determine the M segmented frequencies as passband cutoff frequencies, design M filter triples, with one segmented frequency corresponding to one filter triple, and the filter triple including: a low-pass filter, a high-pass filter, and an all-pass filter; the first bandpass filter includes at least one target filter, and one target filter corresponds to one filter in one filter triple; wherein, when the at least one target filter is at least two target filters, the at least two filter triples corresponding to the at least two target filters are different.
[0150] Optionally, in this embodiment, the processing module 902 is further configured to: design an even-order first low-pass filter corresponding to the first segmentation frequency based on the target design method; design a first all-pass filter based on the group delay of the first low-pass filter; and design a first high-pass filter based on the first low-pass filter and the first all-pass filter; wherein the M segmentation frequencies include the first frequency, and the M filter triplets include a first filter triplet, which includes the first low-pass filter, the first all-pass filter, and the first high-pass filter.
[0151] Optionally, in this embodiment of the application, the processing module 902 is specifically used to: perform cascade processing on the sub-filters in the target bandpass filter bank; input the first audio signal into the cascaded target bandpass filter bank for filtering processing to obtain the second audio signal.
[0152] Optionally, in this embodiment, the processing module 902 is further configured to: design a first auxiliary filter bank based on the Butterworth filter design method, the first auxiliary filter bank including a first auxiliary low-pass filter and a first auxiliary high-pass filter; design a second low-pass filter based on the first auxiliary low-pass filter; design a second high-pass filter based on the first auxiliary high-pass filter; and design a second all-pass filter based on the second low-pass filter and the second high-pass filter; wherein the second filter triplet includes the second low-pass filter, the second all-pass filter, and the second high-pass filter; the second filter triplet is an odd-order filter, the order of the second filter triplet is the same as the order of the first auxiliary filter bank, the second low-pass filter is the first auxiliary low-pass filter, and the second high-pass filter is the first auxiliary high-pass filter or a negative of the first auxiliary high-pass filter.
[0153] Optionally, in this embodiment, the processing module 902 is further configured to: design a second auxiliary filter bank based on the Butterworth filter design method, the second auxiliary filter bank including a second auxiliary low-pass filter and a second auxiliary high-pass filter; design a third low-pass filter based on the second auxiliary low-pass filter; design a third high-pass filter based on the second auxiliary high-pass filter; and design a third all-pass filter based on the third low-pass filter and the third high-pass filter; wherein the third filter triplet includes the third low-pass filter, the third all-pass filter, and the third high-pass filter; the third filter triplet is an even-order filter, the second auxiliary filter bank is an even-order filter, the order of the third filter triplet is twice the order of the second auxiliary filter bank, the third low-pass filter is the square of the second auxiliary low-pass filter, and the third high-pass filter is the square of the second auxiliary high-pass filter.
[0154] Optionally, in this embodiment, the processing module 902 is further configured to: design a third auxiliary filter bank based on the Butterworth filter design method, the third auxiliary filter bank including a third auxiliary low-pass filter and a third auxiliary high-pass filter; design a fourth low-pass filter based on the third auxiliary low-pass filter; design a fourth high-pass filter based on the third auxiliary high-pass filter; and design a fourth all-pass filter based on the fourth low-pass filter and the fourth high-pass filter; wherein the fourth filter triplet includes: the fourth low-pass filter, the fourth all-pass filter, and the fourth high-pass filter; the fourth filter triplet is an even-order filter, the third auxiliary filter bank is an odd-order filter, the order of the fourth filter triplet is twice the order of the third auxiliary filter bank, the fourth low-pass filter is the square of the third auxiliary low-pass filter, and the third high-pass filter is the negative square of the third auxiliary high-pass filter.
[0155] Optionally, in the embodiments of this application, the processing module 902 is further configured to replace the target filter with an all-pass filter in the target filter triplet when the target filter is a high-pass filter in the target filter triplet.
[0156] In the audio signal processing apparatus provided in this application embodiment, a first audio signal is acquired; the first audio signal is then input into a target bandpass filter group for filtering to obtain a second audio signal. The target bandpass filter group includes N first bandpass filters, which are correlated with each other. The second audio signal is then processed to obtain a third audio signal. Wherein, if the target bandpass filter group is a linear-phase FIR bandpass filter group, the first bandpass filters are even-order filters; if the target bandpass filter group is a nonlinear-phase IIR bandpass filter group, the first bandpass filters are designed according to their order. Thus, since the N groups of first bandpass filters in the target bandpass filter group in this application embodiment are correlated rather than independent, a higher quality audio signal is obtained, effectively reducing the loss of sound quality during audio signal processing.
[0157] The audio signal processing device in this application embodiment can be an electronic device or a component within an electronic device, such as an integrated circuit or a chip. The electronic device can be a terminal or other devices besides a terminal. For example, the electronic device can be a mobile phone, tablet computer, laptop computer, PDA, in-vehicle electronic device, mobile internet device (MID), augmented reality (AR) / virtual reality (VR) device, robot, wearable device, ultra-mobile personal computer (UMPC), netbook, or personal digital assistant (PDA), etc. It can also be a server, network attached storage (NAS), personal computer (PC), television set (TV), ATM, or self-service machine, etc. This application embodiment does not specifically limit the scope of the device.
[0158] The audio signal processing device in this application embodiment can be a device with an operating system. This operating system can be Android, iOS, or other possible operating systems; this application embodiment does not specifically limit the specific operating system used.
[0159] The audio signal processing device provided in this application embodiment can achieve... Figure 8 The various processes implemented in the method implementation examples will not be described again here to avoid repetition.
[0160] Optionally, such as Figure 9As shown, this application embodiment also provides an electronic device 1100, including a processor 1101 and a memory 1102. The memory 1102 stores a program or instructions that can run on the processor 1101. When the program or instructions are executed by the processor 1101, they implement the various steps of the above-described audio signal processing method embodiment and can achieve the same technical effect. To avoid repetition, they will not be described again here.
[0161] It should be noted that the electronic devices in the embodiments of this application include the mobile electronic devices and non-mobile electronic devices described above.
[0162] Figure 10 A schematic diagram of the hardware structure of an electronic device to implement an embodiment of this application.
[0163] The electronic device 100 includes, but is not limited to, components such as: radio frequency unit 101, network module 102, audio output unit 103, input unit 104, sensor 105, display unit 106, user input unit 107, interface unit 108, memory 109, and processor 110.
[0164] Those skilled in the art will understand that the electronic device 100 may also include a power supply (such as a battery) for supplying power to various components. The power supply may be logically connected to the processor 110 through a power management system, thereby enabling functions such as managing charging, discharging, and power consumption through the power management system. Figure 10 The electronic device structure shown does not constitute a limitation on the electronic device. The electronic device may include more or fewer components than shown, or combine certain components, or have different component arrangements, which will not be elaborated here.
[0165] The processor 110 is configured to acquire a first audio signal; the processor 110 is configured to input the first audio signal into a target bandpass filter bank for filtering to obtain a second audio signal, the target bandpass filter bank including N first bandpass filters, the N first bandpass filters being correlated with each other; the processor 110 is also configured to perform signal processing on the second audio signal to obtain a third audio signal; wherein, when the target bandpass filter bank is a linear phase FIR bandpass filter bank, the first bandpass filters are even-order filters; the processor 110 is also configured to design a first bandpass filter based on the order of the first bandpass filters when the target bandpass filter bank is a nonlinear phase IIR bandpass filter bank.
[0166] Optionally, in this embodiment of the application, the processor 110 is specifically used to input the first audio signal into a target bandpass filter bank for filtering according to N preset sub-bands to obtain a second audio signal; wherein, the second audio signal includes N sub-band signals corresponding to the N preset sub-bands, and one preset sub-band corresponds to one first bandpass filter.
[0167] Optionally, in this embodiment, the processor 110 is further configured to determine the M segmented frequencies as passband cutoff frequencies, design M filter triples, with one segmented frequency corresponding to one filter triple, and the filter triple including: a low-pass filter, a high-pass filter, and an all-pass filter; the first bandpass filter includes at least one target filter, and one target filter corresponds to one filter in one filter triple; wherein, when the at least one target filter is at least two target filters, the at least two filter triples corresponding to the at least two target filters are different.
[0168] Optionally, in this embodiment of the application, the processor 110 is further configured to: design an even-order first low-pass filter corresponding to the first segmentation frequency based on the target design method; design a first all-pass filter based on the group delay of the first low-pass filter; and design a first high-pass filter based on the first low-pass filter and the first all-pass filter; wherein the M segmentation frequencies include the first frequency, and the M filter triples include a first filter triple, which includes the first low-pass filter, the first all-pass filter, and the first high-pass filter.
[0169] Optionally, in this embodiment of the application, the processor 110 is specifically used to: perform cascade processing on the sub-filters in the target bandpass filter bank; and input the first audio signal into the cascaded target bandpass filter bank for filtering processing to obtain the second audio signal.
[0170] Optionally, in this embodiment, the processor 110 is further configured to: design a first auxiliary filter bank based on a Butterworth filter design method, the first auxiliary filter bank including a first auxiliary low-pass filter and a first auxiliary high-pass filter; design a second low-pass filter based on the first auxiliary low-pass filter; design a second high-pass filter based on the first auxiliary high-pass filter; and design a second all-pass filter based on the second low-pass filter and the second high-pass filter; wherein the second filter triplet includes the second low-pass filter, the second all-pass filter, and the second high-pass filter; the second filter triplet is an odd-order filter, the order of the second filter triplet is the same as the order of the first auxiliary filter bank, the second low-pass filter is the first auxiliary low-pass filter, and the second high-pass filter is the first auxiliary high-pass filter or a negative of the first auxiliary high-pass filter.
[0171] Optionally, in this embodiment, the processor 110 is further configured to: design a second auxiliary filter bank based on a Butterworth filter design method, the second auxiliary filter bank including a second auxiliary low-pass filter and a second auxiliary high-pass filter; design a third low-pass filter based on the second auxiliary low-pass filter; design a third high-pass filter based on the second auxiliary high-pass filter; and design a third all-pass filter based on the third low-pass filter and the third high-pass filter; wherein the third filter triplet includes the third low-pass filter, the third all-pass filter, and the third high-pass filter; the third filter triplet is an even-order filter, the second auxiliary filter bank is an even-order filter, the order of the third filter triplet is twice the order of the second auxiliary filter bank, the third low-pass filter is the square of the second auxiliary low-pass filter, and the third high-pass filter is the square of the second auxiliary high-pass filter.
[0172] Optionally, in this embodiment, the processor 110 is further configured to: design a third auxiliary filter bank based on a Butterworth filter design method, the third auxiliary filter bank including a third auxiliary low-pass filter and a third auxiliary high-pass filter; design a fourth low-pass filter based on the third auxiliary low-pass filter; design a fourth high-pass filter based on the third auxiliary high-pass filter; and design a fourth all-pass filter based on the fourth low-pass filter and the fourth high-pass filter; wherein the fourth filter triplet includes: the fourth low-pass filter, the fourth all-pass filter, and the fourth high-pass filter; the fourth filter triplet is an even-order filter, the third auxiliary filter bank is an odd-order filter, the order of the fourth filter triplet is twice the order of the third auxiliary filter bank, the fourth low-pass filter is the square of the third auxiliary low-pass filter, and the third high-pass filter is the negative square of the third auxiliary high-pass filter.
[0173] Optionally, in this embodiment of the application, the processor 110 is further configured to replace the target filter with an all-pass filter in the target filter triplet when the target filter is a high-pass filter in the target filter triplet.
[0174] In the electronic device provided in this application embodiment, a first audio signal is acquired; and the first audio signal is input into a target bandpass filter group for filtering to obtain a second audio signal. The target bandpass filter group includes N first bandpass filters, and the N first bandpass filters are correlated with each other. The second audio signal is then processed to obtain a third audio signal. Wherein, if the target bandpass filter group is a linear-phase FIR bandpass filter group, the first bandpass filters are even-order filters; if the target bandpass filter group is a nonlinear-phase IIR bandpass filter group, the first bandpass filters are designed according to their order. Thus, since the N groups of first bandpass filters in the target bandpass filter group in this application embodiment are correlated rather than independent, a higher quality audio signal is obtained, effectively reducing the loss of sound quality during audio signal processing.
[0175] It should be understood that, in this embodiment, the input unit 104 may include a graphics processing unit (GPU) 1041 and a microphone 1042. The GPU 1041 processes image data of still images or videos obtained by an image capture device (such as a camera) in video capture mode or image capture mode. The display unit 106 may include a display panel 1061, which may be configured in the form of a liquid crystal display, an organic light-emitting diode, or the like. The user input unit 107 includes at least one of a touch panel 1071 and other input devices 1072. The touch panel 1071 is also called a touch screen. The touch panel 1071 may include a touch detection device and a touch controller. Other input devices 1072 may include, but are not limited to, a physical keyboard, function keys (such as volume control buttons, power buttons, etc.), a trackball, a mouse, and a joystick, which will not be described in detail here.
[0176] The memory 109 can be used to store software programs and various data. The memory 109 may primarily include a first storage area for storing programs or instructions and a second storage area for storing data. The first storage area may store the operating system, application programs or instructions required for at least one function (such as sound playback, image playback, etc.). Furthermore, the memory 109 may include volatile memory or non-volatile memory, or both. The non-volatile memory may be read-only memory (ROM), programmable read-only memory (PROM), erasable programmable read-only memory (EPROM), electrically erasable programmable read-only memory (EEPROM), or flash memory. Volatile memory can be random access memory (RAM), static random access memory (SRAM), dynamic random access memory (DRAM), synchronous dynamic random access memory (SDRAM), double data rate synchronous dynamic random access memory (DDRSDRAM), enhanced synchronous dynamic random access memory (ESDRAM), synchronous link dynamic random access memory (SLDRAM), and direct memory bus RAM (DRRAM). The memory 109 in the embodiments of this application includes, but is not limited to, these and any other suitable types of memory.
[0177] Processor 110 may include one or more processing units; optionally, processor 110 integrates an application processor and a modem processor, wherein the application processor mainly handles operations involving the operating system, user interface, and applications, and the modem processor mainly handles wireless communication signals, such as a baseband processor. It is understood that the aforementioned modem processor may also not be integrated into processor 110.
[0178] This application also provides a readable storage medium storing a program or instructions. When the program or instructions are executed by a processor, they implement the various processes of the above-described audio signal processing method embodiments and achieve the same technical effect. To avoid repetition, they will not be described again here.
[0179] The processor is the processor in the electronic device described in the above embodiments. The readable storage medium includes computer-readable storage media, such as computer read-only memory (ROM), random access memory (RAM), magnetic disk, or optical disk.
[0180] This application embodiment also provides a chip, which includes a processor and a communication interface. The communication interface is coupled to the processor. The processor is used to run programs or instructions to implement the various processes of the above-described audio signal processing method embodiments and can achieve the same technical effect. To avoid repetition, it will not be described again here.
[0181] It should be understood that the chip mentioned in the embodiments of this application may also be referred to as a system-on-a-chip, system chip, chip system, or system-on-a-chip, etc.
[0182] This application provides a computer program product, which is stored in a storage medium and executed by at least one processor to implement the various processes of the audio signal processing method embodiments described above, and can achieve the same technical effect. To avoid repetition, it will not be described again here.
[0183] It should be noted that, in this document, the terms "comprising," "including," or any other variations thereof are intended to cover non-exclusive inclusion, such that a process, method, article, or apparatus that comprises a list of elements includes not only those elements but also other elements not expressly listed, or elements inherent to such a process, method, article, or apparatus. Without further limitations, an element defined by the phrase "comprising one..." does not exclude the presence of other identical elements in the process, method, article, or apparatus that includes that element. Furthermore, it should be noted that the scope of the methods and apparatuses in the embodiments of this application is not limited to performing functions in the order shown or discussed, but may also include performing functions substantially simultaneously or in the reverse order, depending on the functions involved. For example, the described methods may be performed in a different order than described, and various steps may be added, omitted, or combined. Additionally, features described with reference to certain examples may be combined in other examples.
[0184] Through the above description of the embodiments, those skilled in the art can clearly understand that the methods of the above embodiments can be implemented by means of software plus necessary general-purpose hardware platforms. Of course, they can also be implemented by hardware, but in many cases the former is a better implementation method. Based on this understanding, the technical solution of this application, in essence, or the part that contributes to the prior art, can be embodied in the form of a computer software product. This computer software product is stored in a storage medium (such as ROM / RAM, magnetic disk, optical disk) and includes several instructions to cause a terminal (which may be a mobile phone, computer, server, or network device, etc.) to execute the methods described in the various embodiments of this application.
[0185] The embodiments of this application have been described above with reference to the accompanying drawings. However, this application is not limited to the specific embodiments described above. The specific embodiments described above are merely illustrative and not restrictive. Those skilled in the art can make many other forms under the guidance of this application without departing from the spirit and scope of the claims, and all of these forms are within the protection scope of this application.
Claims
1. An audio signal processing method, characterized in that, The method includes: Acquire the first audio signal; The first audio signal is input into a target bandpass filter bank for filtering to obtain a second audio signal. The target bandpass filter bank includes N first bandpass filters. The N first bandpass filters are associated with M segmented frequencies. Each first bandpass filter is composed of at least one filter from a filter triplet designed with the M segmented frequencies as passband cutoff frequencies. The filter triplet corresponds one-to-one with the segmented frequencies and includes a low-pass filter, a high-pass filter, and an all-pass filter. M=N-1. The second audio signal is processed to obtain the third audio signal; Wherein, when the target bandpass filter bank is a linear phase FIR bandpass filter bank, the first bandpass filter is an even-order filter; When the target bandpass filter bank is a nonlinear phase IIR bandpass filter bank, the first bandpass filter is designed according to the order of the first bandpass filter.
2. The method according to claim 1, characterized in that, The step of inputting the first audio signal into a target bandpass filter bank for filtering to obtain the second audio signal includes: The first audio signal is input into the target bandpass filter bank for filtering according to N preset sub-bands to obtain the second audio signal; The second audio signal includes N sub-band signals corresponding to the N preset sub-bands, and one preset sub-band corresponds to one first bandpass filter.
3. The method according to claim 2, characterized in that, The N preset sub-bands correspond to M segmented frequencies. Before inputting the first audio signal into the target bandpass filter bank for filtering to obtain the second audio signal, the method further includes: The M segmented frequencies are determined as passband cutoff frequencies, and M filter triples are designed. Each segmented frequency corresponds to one filter triple, and the filter triple includes: a low-pass filter, a high-pass filter, and an all-pass filter. The first bandpass filter includes at least one target filter, and one target filter corresponds to one filter in a filter triplet; Wherein, when the at least one target filter is at least two target filters, the at least two filter triples corresponding to the at least two target filters are different.
4. The method according to claim 3, characterized in that, The target bandpass filter bank is a linear-phase FIR bandpass filter bank, and the method further includes: Design an even-order first low-pass filter corresponding to the first segmentation frequency based on the target design method; Based on the group delay of the first low-pass filter, design the first all-pass filter; Based on the first low-pass filter and the first full-pass filter, design a first high-pass filter; The M segmented frequencies include a first frequency, and the M filter triples include a first filter triple, which includes a first low-pass filter, a first full-pass filter, and a first high-pass filter.
5. The method according to claim 4, characterized in that, The step of inputting the first audio signal into a target bandpass filter bank for filtering to obtain the second audio signal includes: The sub-filters in the target bandpass filter bank are cascaded. The first audio signal is input into the target bandpass filter bank after being processed in series to filter it, thereby obtaining the second audio signal.
6. The method according to claim 3, characterized in that, The target bandpass filter bank is a nonlinear phase IIR bandpass filter bank, the M filter triplets include a second filter triplet, and the method further includes: The first auxiliary filter bank is designed based on the Butterworth filter design method. The first auxiliary filter bank includes a first auxiliary low-pass filter and a first auxiliary high-pass filter. Based on the first auxiliary low-pass filter, design a second low-pass filter; Based on the first auxiliary high-pass filter, design a second high-pass filter; Based on the second low-pass filter and the second high-pass filter, design a second all-pass filter; The second filter triplet includes: the second low-pass filter, the second full-pass filter, and the second high-pass filter; The second filter triplet is an odd-order filter, and the order of the second filter triplet is the same as the order of the first auxiliary filter group. The second low-pass filter is the first auxiliary low-pass filter, and the second high-pass filter is the first auxiliary high-pass filter or the negative of the first auxiliary high-pass filter.
7. The method according to claim 3, characterized in that, The target bandpass filter bank is a nonlinear phase IIR bandpass filter bank, the M filter triplets include a third filter triplet, and the method further includes: A second auxiliary filter bank is designed based on the Butterworth filter design method. The second auxiliary filter bank includes a second auxiliary low-pass filter and a second auxiliary high-pass filter. Based on the second auxiliary low-pass filter, design a third low-pass filter; Based on the second auxiliary high-pass filter, design a third high-pass filter; Based on the third low-pass filter and the third high-pass filter, design a third full-pass filter; The third filter triplet includes: the third low-pass filter, the third full-pass filter, and the third high-pass filter; The third filter triplet is an even-order filter, the second auxiliary filter group is an even-order filter, the order of the third filter triplet is twice the order of the second auxiliary filter group, the third low-pass filter is the square of the second auxiliary low-pass filter, and the third high-pass filter is the square of the second auxiliary high-pass filter.
8. The method according to claim 3, characterized in that, The target bandpass filter bank is a nonlinear phase IIR bandpass filter bank, the M filter triplets include a fourth filter triplet, and the method further includes: A third auxiliary filter bank is designed based on the Butterworth filter design method. The third auxiliary filter bank includes a third auxiliary low-pass filter and a third auxiliary high-pass filter. Based on the third auxiliary low-pass filter, design a fourth low-pass filter; Based on the third auxiliary high-pass filter, design a fourth high-pass filter; Based on the fourth low-pass filter and the fourth high-pass filter, design a fourth full-pass filter; The fourth filter triplet includes: the fourth low-pass filter, the fourth full-pass filter, and the fourth high-pass filter; The fourth filter triplet is an even-order filter, the third auxiliary filter group is an odd-order filter, the order of the fourth filter triplet is twice the order of the third auxiliary filter group, the fourth low-pass filter is the square of the third auxiliary low-pass filter, and the fourth high-pass filter is the negative square of the third auxiliary high-pass filter.
9. The method according to any one of claims 6-8, characterized in that, The method further includes: If the target filter is a high-pass filter in the target filter triplet, the target filter is replaced with an all-pass filter in the target filter triplet.
10. An audio signal processing device, characterized in that, The device includes: an acquisition module and a processing module; The acquisition module is used to acquire the first audio signal; The processing module is used to input the first audio signal acquired by the acquisition module into a target bandpass filter group for filtering to obtain a second audio signal. The target bandpass filter group includes N first bandpass filters. The N first bandpass filters are associated with M segmented frequencies. Each first bandpass filter is composed of at least one filter in a filter triplet designed with the M segmented frequencies as passband cutoff frequencies. The filter triplet corresponds one-to-one with the segmented frequencies and includes a low-pass filter, a high-pass filter, and an all-pass filter. M=N-1. The processing module is further configured to perform signal processing on the second audio signal to obtain a third audio signal; Wherein, when the target bandpass filter bank is a linear phase FIR bandpass filter bank, the first bandpass filter is an even-order filter; The processing module is further configured to design the first bandpass filter based on the order of the first bandpass filter when the target bandpass filter bank is a nonlinear phase IIR bandpass filter bank.
11. The apparatus according to claim 10, characterized in that, The processing module is specifically used to input the first audio signal acquired by the acquisition module into the target bandpass filter bank for filtering processing according to N preset sub-bands to obtain the second audio signal; The second audio signal includes N sub-band signals corresponding to the N preset sub-bands, and one preset sub-band corresponds to one first bandpass filter.
12. The apparatus according to claim 11, characterized in that, The processing module is also used to determine the M segmented frequencies as passband cutoff frequencies and design M filter triples, with one segmented frequency corresponding to one filter triple. The filter triple includes a low-pass filter, a high-pass filter, and an all-pass filter. The first bandpass filter includes at least one target filter, and one target filter corresponds to one filter in a filter triplet; Wherein, when the at least one target filter is at least two target filters, the at least two filter triples corresponding to the at least two target filters are different.
13. An electronic device, characterized in that, It includes a processor, a memory, and a program or instructions stored in the memory and executable on the processor, wherein the program or instructions, when executed by the processor, implement the steps of the audio signal processing method as described in any one of claims 1 to 9.
14. A readable storage medium, characterized in that, The readable storage medium stores a program or instructions that, when executed by a processor, implement the steps of the audio signal processing method as described in any one of claims 1 to 9.