Filtering method, device and equipment of segmented audio signal and storage medium

Abstract

The application belongs to the technical field of signal processing, and discloses a filtering method, device and equipment for segmented audio signals and a storage medium. The method comprises the following steps: constructing a target filter according to a preset filter type, a preset frequency division point and a preset filter order; reading current segmented audio signals received by each filter channel; obtaining splicing sampling points of initial segmented audio signals corresponding to each filter channel, splicing the splicing sampling points of each filter channel with the current segmented audio signals, and obtaining splicing segmented audio signals corresponding to each filter channel; obtaining filter segmented audio signals corresponding to each filter channel according to the splicing segmented audio signals and the target filter; superimposing the filter segmented audio signals corresponding to each filter channel to obtain a target segmented audio signal, and inputting the target segmented audio signal into a target channel. In the above manner, the listening effect of the segmented audio signals after filtering is improved without losing the audio signals.

Description

Technical Field

[0001] This invention relates to the field of signal processing technology, and in particular to a method, apparatus, device, and storage medium for filtering segmented audio signals. Background Technology

[0002] Audio signal filtering simulation typically uses an offline mode. First, the entire audio segment is read locally, then filtered, and the filtered signal is saved. Finally, the filtered audio segment is read and output through an external speaker. After filtering, although there is a certain delay, the signal is continuous and does not affect the listening experience. In real-time audio signal transmission processing, audio is first played on a mobile music app. The audio signal is transmitted from the phone to the computer via Bluetooth. The real-time audio signal is filtered and output through an external speaker. During this process, segmented audio signals need to be filtered. Due to the delay caused by filtering, each segment is discontinuous, resulting in a poor listening experience. To improve the listening experience after filtering segmented audio signals, the delayed portion of each segment is usually truncated. While this improves the listening experience to some extent, it also results in the loss of some audio signal.

[0003] The content described herein is for the purpose of assisting in understanding the technical solution of the present invention and does not imply an admission that the content is prior art. Summary of the Invention

[0004] The main objective of this invention is to provide a method, apparatus, device, and storage medium for filtering segmented audio signals, aiming to solve the technical problem in the existing technology of real-time transmission and processing of audio signals, which results in the loss of some audio signals in order to improve the listening effect after filtering segmented audio signals.

[0005] To achieve the aforementioned objective, the present invention provides a filtering method for segmented audio signals, the filtering method comprising the following steps:

[0006] Construct the target filter according to the preset requirements;

[0007] Read the current segmented audio signal received from each filter channel;

[0008] Obtain the splicing sampling points of the initial segmented audio signal corresponding to each filter channel, and splice the splicing sampling points of each filter channel with the current segmented audio signal to obtain the spliced ​​segmented audio signal corresponding to each filter channel;

[0009] Based on the spliced ​​segmented audio signals corresponding to each filtering channel and the target filter, the filtered segmented audio signals corresponding to each filtering channel are obtained, and the length of the filtered segmented audio signals is equal to the length of the current segmented audio signal;

[0010] The filtered segmented audio signals corresponding to each filtering channel are superimposed to obtain the target segmented audio signal, and the target segmented audio signal is input into the target channel.

[0011] Optionally, constructing the target filter according to preset requirements includes:

[0012] Based on the preset requirements, determine the filter type, crossover point, filter order, and filter constructor.

[0013] Input the filter type, the frequency division point, and the filter order into the filter constructor to determine the filter coefficients;

[0014] The target filter is obtained based on the filtering coefficients, and the target filter is either a finite impulse response filter or an infinite impulse response filter.

[0015] Optionally, obtaining the filtered segmented audio signal corresponding to each filtered channel based on the spliced ​​segmented audio signal corresponding to each filtered channel and the target filter includes:

[0016] When the target filter is a finite impulse response filter, the spliced ​​segmented audio signal corresponding to each filter channel is used as the input signal;

[0017] The input signal is input into the finite impulse response filter to obtain the filtered segmented audio signal corresponding to each filtering channel.

[0018] Optionally, before reading the currently segmented audio signal received by each filter channel, the method further includes:

[0019] The number of splicing sampling points is determined based on the filter order;

[0020] Based on the number of splicing sampling points and the preset sampling positions, the splicing sampling points are determined in the initial segmented audio signals corresponding to each filtered channel.

[0021] Optionally, obtaining the filtered segmented audio signal corresponding to each filtered channel based on the spliced ​​segmented audio signal corresponding to each filtered channel and the target filter includes:

[0022] When the target filter is an infinite impulse response filter, the historical filtered audio signals corresponding to each filter channel are acquired;

[0023] The spliced ​​segmented audio signal corresponding to each filtering channel is used as the first input signal, and the historical filtered audio signal corresponding to each filtering channel is used as the second input signal.

[0024] The first input signal and the second input signal are input into the infinite impulse response filter to obtain the filtered segmented audio signal.

[0025] Optionally, the filter order includes a first filter order and a second filter order, and before reading the currently segmented audio signal received from each filter channel, the method further includes:

[0026] The number of splicing sampling points is determined based on the order of the first filter;

[0027] Based on the number of splicing sampling points and the preset sampling positions, the splicing sampling points are determined in the initial segmented audio signals corresponding to each filtered channel;

[0028] The number of historical filtered audio signal sampling points is determined based on the second filter order;

[0029] Based on the number of sampling points of the historical filtered audio signal, the historical filtered audio signal corresponding to each filtered channel is determined.

[0030] Optionally, after superimposing the filtered segmented audio signals corresponding to each filtered channel to obtain the target segmented audio signal, and inputting the target segmented audio signal into the target channel, the method further includes:

[0031] The current segmented audio signal is used as the new initial segmented audio signal. Based on the new initial segmented audio signal, the splicing sampling points corresponding to each filter channel are determined and retained. Then, the process returns to the step of reading the current segmented audio signal received by each filter channel.

[0032] Furthermore, to achieve the aforementioned objective, the present invention also proposes a filtering device for segmented audio signals, the filtering device comprising:

[0033] The construction module is used to construct the target filter based on the preset filter type, preset frequency division point, and preset filter order.

[0034] The splicing module is used to read the currently segmented audio signal received from each filter channel;

[0035] The splicing module is also used to obtain the splicing sampling points of the initial segmented audio signals corresponding to each filter channel, and splice the splicing sampling points of each filter channel with the current segmented audio signal to obtain the spliced ​​segmented audio signals corresponding to each filter channel.

[0036] The filtering module is used to obtain the filtered segmented audio signal corresponding to each filtering channel based on the spliced ​​segmented audio signal corresponding to each filtering channel and the target filter, wherein the length of the filtered segmented audio signal is equal to the length of the current segmented audio signal;

[0037] The filtering module is also used to superimpose the filtered segmented audio signals corresponding to each filtering channel to obtain the target segmented audio signal, and input the target segmented audio signal into the target channel.

[0038] Furthermore, to achieve the aforementioned objective, the present invention also proposes a segmented audio signal filtering device, the segmented audio signal filtering device comprising: a memory, a processor, and a segmented audio signal filtering program stored in the memory and executable on the processor, the segmented audio signal filtering program being configured to implement the steps of the segmented audio signal filtering method described above.

[0039] Furthermore, to achieve the aforementioned objective, the present invention also proposes a storage medium storing a filtering program for segmented audio signals, wherein when the filtering program for segmented audio signals is executed by a processor, it implements the steps of the filtering method for segmented audio signals as described above.

[0040] In this invention, a target filter is constructed based on a preset filter type, preset crossover point, and preset filter order. The current segmented audio signal received by each filter channel is read, and the splicing sampling points of the initial segmented audio signal corresponding to each filter channel are obtained. These splicing sampling points are then spliced ​​with the current segmented audio signal to obtain the spliced ​​segmented audio signal corresponding to each filter channel. Based on the spliced ​​segmented audio signal and the target filter, the filtered segmented audio signal corresponding to each filter channel is obtained. These filtered segmented audio signals are then superimposed to obtain the target segmented audio signal, which is then input into the target channel. Since the delayed portion of each audio signal is usually truncated in real-time audio signal transmission processing to improve the listening effect after filtering the segmented audio signal, this results in the loss of some audio signal. This invention improves upon the traditional filtering method by splicing the segmented audio signal with the sampling points of the last part of the previous audio signal before filtering. This results in a segmented audio signal filtering result consistent with the filtering result of the entire audio signal, improving the listening effect of the filtered segmented audio signal without losing the audio signal. Attached Figure Description

[0041] Figure 1 This is a schematic diagram of the structure of a segmented audio signal filtering device in the hardware operating environment involved in the embodiments of the present invention;

[0042] Figure 2 This is a flowchart illustrating the first embodiment of the segmented audio signal filtering method of the present invention;

[0043] Figure 3 This is a flowchart illustrating a second embodiment of the segmented audio signal filtering method of the present invention;

[0044] Figure 4 This is a flowchart illustrating the third embodiment of the segmented audio signal filtering method of the present invention;

[0045] Figure 5 This is a structural block diagram of the first embodiment of the segmented audio signal filtering device of the present invention.

[0046] The realization of the objective, functional features and advantages of the present invention will be further explained in conjunction with the embodiments and with reference to the accompanying drawings. Detailed Implementation

[0047] It should be understood that the specific embodiments described herein are for illustrative purposes only and are not intended to limit the scope of the invention.

[0048] Reference Figure 1 , Figure 1 This is a schematic diagram of the structure of a segmented audio signal filtering device in the hardware operating environment involved in the embodiments of the present invention.

[0049] like Figure 1 As shown, the filtering device for the segmented audio signal may include: a processor 1001, such as a central processing unit (CPU), a communication bus 1002, a user interface 1003, a network interface 1004, and a memory 1005. The communication bus 1002 is used to enable communication between these components. The user interface 1003 may include a display screen or an input unit such as a keyboard; optionally, the user interface 1003 may also include a standard wired interface or a wireless interface. The network interface 1004 may optionally include a standard wired interface or a wireless interface (such as a Wireless-Fidelity (Wi-Fi) interface). The memory 1005 may be a high-speed random access memory (RAM) or a stable non-volatile memory (NVM), such as a disk drive. The memory 1005 may also optionally be a storage device independent of the aforementioned processor 1001.

[0050] Those skilled in the art will understand that Figure 1 The structure shown does not constitute a limitation on the filtering device for segmented audio signals and may include more or fewer components than shown, or combine certain components, or have different component arrangements.

[0051] like Figure 1 As shown, the memory 1005, which serves as a storage medium, may include an operating system, a network communication module, a user interface module, and a filtering program for segmented audio signals.

[0052] exist Figure 1 In the segmented audio signal filtering device shown, the network interface 1004 is mainly used for data communication with the network server; the user interface 1003 is mainly used for data interaction with the user; the processor 1001 and the memory 1005 in the segmented audio signal filtering device of the present invention can be set in the segmented audio signal filtering device, and the segmented audio signal filtering device calls the segmented audio signal filtering program stored in the memory 1005 through the processor 1001 and executes the segmented audio signal filtering method provided in the embodiment of the present invention.

[0053] This invention provides a filtering method for segmented audio signals, referring to... Figure 2 , Figure 2 This is a flowchart illustrating the first embodiment of a segmented audio signal filtering method according to the present invention.

[0054] In this embodiment, the filtering method for the segmented audio signal includes the following steps:

[0055] Step S10: Construct the target filter according to the preset requirements.

[0056] It should be noted that the execution subject of this embodiment is a computer, which is equipped with a segmented audio signal filtering program. By running the segmented audio signal filtering program, each segment of audio signal is filtered during the real-time transmission of audio signals to improve the listening effect of the segmented audio signal after filtering.

[0057] Further, step S10 includes: determining the filter type, crossover point, filter order, and filter constructor according to the preset requirements; inputting the filter type, crossover point, and filter order into the filter constructor to determine the filter coefficients; and obtaining the target filter based on the filter coefficients.

[0058] It is understood that the preset requirements refer to the current requirements, which can be determined according to the actual situation, and this embodiment does not impose any restrictions on them. The filter type refers to the type of filter currently used, which can be either a high-pass filter or a low-pass filter. It can be either a low-pass filter or a high-pass filter, and can be determined according to the actual requirements. The crossover point refers to the crossover point of the filter currently used, for example, 120Hz, and can be determined according to the actual requirements. The filter order refers to the order of the filter currently used, for example, 15, 31, 63, or 127. The higher the filter order, the higher the filter accuracy, but the more computation is required. It needs to be determined according to the actual requirements, and this embodiment does not impose any restrictions on it. The filter constructor refers to the function used to construct the filter. The filter constructor can be either a finite impulse response filter constructor or an infinite impulse response filter constructor. It can be the fir1 function corresponding to the finite impulse response filter or the function corresponding to the infinite impulse response filter, and can be determined according to the actual requirements. This embodiment does not impose any restrictions on it. By inputting the filter type, crossover point, and filter order into the filter constructor, the filter coefficients, i.e., the filter coefficients, can be obtained. Different filter constructors correspond to different filter coefficients.

[0059] It should be understood that the target filter refers to the filter that needs to be used. The target filter can be either a finite impulse response (FIR) filter or an infinite impulse response (IOR) filter. If the filter constructor is the same as the constructor for a finite impulse response filter, then the target filter is a finite impulse response filter; if the filter constructor is the same as the constructor for an IOR filter, then the target filter is an IOR filter. After determining the filter coefficients through the constructor, the target filter can be obtained.

[0060] In practical implementation, if the preset requirement is to improve the listening effect while enriching low-frequency information, the filter type can be set to low-pass filter, the crossover point can be set to 120Hz, and the filter order can be set to N. Then, an appropriate constructor can be selected according to the actual needs to construct an N-order low-pass filter with a crossover point of 120Hz for subsequent audio signal filtering.

[0061] Step S20: Read the current segmented audio signal received by each filter channel.

[0062] It should be noted that the filtering channel refers to the channel that needs to be filtered. 7.1.4 The panoramic sound system is a system with a total of 12 channels, consisting of 7 front, rear, left, and right surround speakers, 1 subwoofer, and 4 overhead speakers. If richer low-frequency information is needed, the audio signals of the other 11 channels (excluding the subwoofer channel) can be filtered and then input into the subwoofer channel. At this time, the filtered channels are the other 11 channels, namely the 7 front, rear, left, and right surround speaker channels and the 4 overhead speaker channels.

[0063] It is understandable that during real-time audio signal transmission, filtering is performed in segments. The current segment of the audio signal refers to the currently received audio signal. The length of the current segment needs to be determined based on the actual situation, usually 1024. This embodiment does not impose this limitation and can use K to represent the length of the current segment. Each channel has a corresponding current segment of the audio signal.

[0064] In the specific implementation, the current segmented audio signal of length K in each filter channel is received and read in real time.

[0065] Step S30: Obtain the splicing sampling points of the initial segmented audio signals corresponding to each filter channel, and splice the splicing sampling points of each filter channel with the current segmented audio signal to obtain the spliced ​​segmented audio signals corresponding to each filter channel.

[0066] It should be understood that the initial segmented audio signal refers to the previously received audio signal, i.e., the previous audio segment. For the first audio segment, it can be assumed that the splicing sampling point value of its previous audio segment is 0. The splicing sampling point refers to the sampling points in the previous audio segment that need to be spliced ​​with the current segmented audio signal. These are usually the last few sampling points, and the number of splicing sampling points needs to be determined based on the order of the target filter. The spliced ​​segmented audio signal refers to the spliced ​​audio signal. Each channel has a corresponding previous audio segment, thus each channel can determine its corresponding splicing sampling points, thereby obtaining the spliced ​​segmented audio signal for each channel. In each spliced ​​segmented audio signal, the splicing sampling point is spliced ​​before the current segmented audio signal. For example, if the splicing sampling point is A and the current segmented audio signal is B, then the spliced ​​segmented audio signal is A+B.

[0067] It should be noted that when filtering segmented audio signals, each segment has a delay caused by the filtering, resulting in discontinuity between the filtered segments and a poor listening experience. In this embodiment, the sampling points of the last part of the previous audio segment are spliced ​​into the current segmented audio signal to ensure that the filtered signal is also continuous and improve the listening experience.

[0068] In the specific implementation, the splicing sampling points of each filter channel in the previous audio signal are spliced ​​with the current segment audio signal to obtain the spliced ​​segment audio signal corresponding to each filter channel.

[0069] Step S40: Based on the spliced ​​segmented audio signals corresponding to each filtering channel and the target filter, obtain the filtered segmented audio signals corresponding to each filtering channel. The length of the filtered segmented audio signals is equal to the length of the current segmented audio signal.

[0070] It is understood that the filtered segmented audio signal refers to the filtered signal. Through the scheme described in this embodiment, the length of the filtered audio signal is equal to the length of the current segmented audio signal.

[0071] In the actual implementation, the obtained spliced ​​segmented audio signals are filtered to obtain the filtered audio signals of each filter channel.

[0072] Step S50: Superimpose the filtered segmented audio signals corresponding to each filtered channel to obtain the target segmented audio signal, and input the target segmented audio signal into the target channel.

[0073] It should be understood that the target segmented audio signal refers to the audio signal that ultimately needs to be input into the target channel. The target channel is the channel that can be improved at present. It can be a low-frequency channel or a high-frequency channel, which needs to be determined according to the actual situation. In the 7.1.4 panoramic sound system, if it is necessary to enrich the low-frequency information, the audio signals of the other 11 channels except the subwoofer channel can be filtered and then input into the subwoofer channel. At this time, the subwoofer channel can be used as the target channel. In other embodiments, the high-frequency channel can also be selected as the target channel to enrich the high-frequency information.

[0074] In the specific implementation, during the dual-channel stereo to 7.1.4 Dolby conversion process, the audio signals of the 11 channels other than the subwoofer channel can be low-pass filtered and superimposed before being input into the subwoofer channel to enrich the low-frequency information.

[0075] Furthermore, after step S50, the method further includes: using the current segmented audio signal as a new initial segmented audio signal, determining and retaining the splicing sampling points corresponding to each filter channel based on the new initial segmented audio signal, and returning to the step of reading the current segmented audio signal received by each filter channel.

[0076] In the actual implementation, the splicing sampling points of the current segmented audio signal corresponding to each filtering channel are retained for use in the next segment of audio signal.

[0077] In this embodiment, a target filter is constructed based on a preset filter type, preset frequency division point, and preset filter order. The current segmented audio signal received by each filter channel is read, and the splicing sampling points of the initial segmented audio signal corresponding to each filter channel are obtained. The splicing sampling points of each filter channel are spliced ​​with the current segmented audio signal to obtain the spliced ​​segmented audio signal corresponding to each filter channel. Based on the spliced ​​segmented audio signal and the target filter, the filtered segmented audio signal corresponding to each filter channel is obtained. The filtered segmented audio signals corresponding to each filter channel are superimposed to obtain the target segmented audio signal, which is then input into the target channel. In real-time audio signal transmission and processing, the delayed portion of each audio signal is usually truncated to improve the listening effect after segmented audio signal filtering. However, this results in the loss of some audio signal. This embodiment improves the traditional filtering method by concatenating the segmented audio signal with the sampling points of the last part of the previous audio signal before filtering. This results in a segmented audio signal filtering result that is consistent with the filtering result of the whole audio signal. This improves the listening effect of the segmented audio signal filtering without losing the audio signal. In addition, the low-frequency / high-frequency information of the channel can be enriched according to actual needs.

[0078] Reference Figure 3 , Figure 3 This is a flowchart illustrating a second embodiment of a segmented audio signal filtering method according to the present invention.

[0079] Based on the above embodiments, step S40 includes:

[0080] Step S401: When the target filter is a finite impulse response filter, the spliced ​​segmented audio signal corresponding to each filter channel is used as the input signal.

[0081] It should be noted that the finite impulse response filter can be expressed by the following formula:

[0082]

[0083] In the formula, y represents the output signal, x represents the input signal, a represents the filter coefficient, K represents the filter order, n represents the signal sampling point index, and k represents the filter coefficient index.

[0084] It is understandable that, according to the above formula, the current output signal y(n) is the weighted sum of the current input signal x(n) and the previous N input signals. When filtering an audio signal, a complete historical input signal is only available starting from the (N+1)th signal sampling point. The current output signal y(N+1) can be obtained by calculating the weighted sum of the current input signal x(N+1) and the previous N input signals. If filtering is applied to the entire audio signal, then only the first N signal sampling points of the entire audio signal lack a complete historical input signal. However, if filtering is applied to segmented audio signals, then the first N sampling points of each segment of the audio signal lack a complete historical input signal. Therefore, in this embodiment, N sampling points are concatenated before the current segmented audio signal so that each segment of the current audio signal of length L can have a complete historical input signal, and an audio signal of length L can be output after filtering.

[0085] In the specific implementation, when filtering segmented audio signals, the input signal of the finite impulse response filter is a signal of length (n+L) formed by splicing the last N sampling points of the previous audio signal and the current segmented audio signal of length L. For the first audio signal, it is assumed that the last N sampling points of the previous audio signal have a value of 0.

[0086] Step S402: Input the input signal into the finite impulse response filter to obtain the filtered segmented audio signal corresponding to each filtering channel.

[0087] In the actual implementation, the input signal of length (N+L) is filtered and output as a filtered segmented audio signal of length L.

[0088] Furthermore, prior to step S20, the procedure also includes:

[0089] Step S101: Determine the number of splicing sampling points based on the filter order.

[0090] It should be understood that the number of spliced ​​sampling points refers to the number of spliced ​​sampling points, which is the same as the filter order. If the filter order is N, then the number of spliced ​​sampling points is N.

[0091] Step S102: Based on the number of splicing sampling points and the preset sampling position, determine the splicing sampling points in the initial segmented audio signals corresponding to each filter channel.

[0092] It should be noted that the preset sampling position refers to the position of the splicing sampling point in the audio signal, which is usually the last part of the previous audio signal.

[0093] In the specific implementation, the last N sampling points in the previous segment of the audio signal of each filtered channel are determined as the splicing sampling points.

[0094] In this embodiment, a finite impulse response (FIR) filter is constructed to filter a segmented audio signal of length L. The input is a signal of length (N+L) formed by concatenating the last N samples of the previous audio segment with the current segment of length L. The output is the filtered audio signal of length L. This embodiment improves upon traditional filtering methods by concatenating the segmented audio signal with the last sample of the previous audio segment before filtering. This results in a segmented audio signal filtering result consistent with the filtered result of the entire audio signal, improving the listening experience of the filtered segmented audio signal without loss of audio signal quality.

[0095] Reference Figure 4 , Figure 4 This is a flowchart illustrating a third embodiment of a segmented audio signal filtering method according to the present invention.

[0096] Based on the above embodiments, step S40 includes:

[0097] Step S401': When the target filter is an infinite impulse response filter, acquire the historical filtered audio signal corresponding to each filter channel, use the spliced ​​segmented audio signal corresponding to each filter channel as the first input signal, and use the historical filtered audio signal corresponding to each filter channel as the second input signal.

[0098] It should be noted that the infinite impulse response filter can be represented by the following formula:

[0099]

[0100] In the formula, y represents the output signal, x represents the input signal, a and b represent filter coefficients, N and P represent filter orders, n represents the signal sampling point index, and k and j represent filter coefficient indices. From the above formula, we can see that the current output signal y(n) is the weighted sum of the current input signal x(n) and the previous N input signals, plus the weighted sum of the previous P output signals.

[0101] It is understandable that the first input signal and the second input signal refer to the two input signals when using an infinite impulse response filter. The first input signal is a signal of length (N+L) formed by concatenating the last N sample points of the previous audio signal and the current audio signal of length L. N sample points are concatenated before the current audio signal to ensure that the current audio signal of length L has a complete historical input signal. The second input signal is the P sample points output previously.

[0102] Step S402': Input the first input signal and the second input signal into the infinite impulse response filter to obtain the filtered segmented audio signal.

[0103] In the specific implementation, the first input signal and the second input signal are filtered and output as a filtered segmented audio signal of length L.

[0104] Furthermore, prior to step S20, the procedure also includes:

[0105] Step S101': Determine the number of splicing sampling points based on the first filter order, and determine the splicing sampling points in the initial segmented audio signals corresponding to each filter channel based on the number of splicing sampling points and the preset sampling position.

[0106] It should be understood that the infinite impulse response filter has two filter orders, N and P, which are the first filter order and the second filter order, respectively. The number of spliced ​​sampling points refers to the number of spliced ​​sampling points, which is the same as the first filter order. If the first filter order is N, then the number of spliced ​​sampling points is N.

[0107] In the specific implementation, the last N sampling points in the previous segment of the audio signal of each filtered channel are determined as the splicing sampling points.

[0108] Step S102': Determine the number of historical filtered audio signal sampling points based on the second filter order, and determine the historical filtered audio signal corresponding to each filtering channel based on the number of historical filtered audio signal sampling points.

[0109] It should be noted that the number of historical filtered audio signal sampling points refers to the number of previously output filtered audio signal sampling points, which is usually equal to the order of the second filter. If the order of the second filter is P, then the number of historical filtered audio signal sampling points is P. The historical filtered audio signal refers to the P previously output sampling points.

[0110] In the specific implementation, the P previously output sampling points are used as the historical filtered audio signals corresponding to each filtering channel for subsequent filtering input.

[0111] In this embodiment, an infinite impulse response filter is used to filter segmented audio signals of length L. The first input signal is a signal of length (N+L) formed by concatenating the last N samples of the previous audio signal with the current segment of length L. The second input signal is the previously output P samples. The output is the filtered audio signal of length L. This embodiment improves upon traditional filtering methods by concatenating the segmented audio signal with the last sample of the previous audio signal before filtering. This results in a segmented audio signal filtering result consistent with the filtered result of the entire audio signal, improving the listening experience of the filtered segmented audio signal without loss of audio signal quality.

[0112] Reference Figure 5 , Figure 5 This is a structural block diagram of the first embodiment of the segmented audio signal filtering device of the present invention.

[0113] like Figure 5 As shown, the segmented audio signal filtering device proposed in this embodiment of the invention includes:

[0114] Module 10 is used to construct a target filter based on a preset filter type, a preset frequency division point, and a preset filter order.

[0115] The splicing module 20 is used to read the currently segmented audio signal received by each filter channel.

[0116] The splicing module 20 is also used to obtain the splicing sampling points of the initial segmented audio signals corresponding to each filter channel, and splice the splicing sampling points of each filter channel with the current segmented audio signal to obtain the spliced ​​segmented audio signals corresponding to each filter channel.

[0117] The filtering module 30 is used to obtain the filtered segmented audio signal corresponding to each filtering channel based on the spliced ​​segmented audio signal corresponding to each filtering channel and the target filter, wherein the length of the filtered segmented audio signal is equal to the length of the current segmented audio signal.

[0118] The filtering module 30 is further configured to superimpose the filtered segmented audio signals corresponding to each filtering channel to obtain a target segmented audio signal, and input the target segmented audio signal into the target channel.

[0119] In this embodiment, a target filter is constructed based on a preset filter type, preset frequency division point, and preset filter order. The current segmented audio signal received by each filter channel is read, and the splicing sampling points of the initial segmented audio signal corresponding to each filter channel are obtained. The splicing sampling points of each filter channel are spliced ​​with the current segmented audio signal to obtain the spliced ​​segmented audio signal corresponding to each filter channel. Based on the spliced ​​segmented audio signal and the target filter, the filtered segmented audio signal corresponding to each filter channel is obtained. The filtered segmented audio signals corresponding to each filter channel are superimposed to obtain the target segmented audio signal, which is then input into the target channel. In real-time audio signal transmission and processing, the delayed portion of each audio signal is usually truncated to improve the listening effect after segmented audio signal filtering. However, this results in the loss of some audio signal. This embodiment improves the traditional filtering method by concatenating the segmented audio signal with the sampling points of the last part of the previous audio signal before filtering. This results in a segmented audio signal filtering result that is consistent with the filtering result of the whole audio signal. This improves the listening effect of the segmented audio signal filtering without losing the audio signal. In addition, the low-frequency / high-frequency information of the channel can be enriched according to actual needs.

[0120] In one embodiment, the construction module 10 is further configured to determine the filter type, frequency division point, filter order, and filter constructor according to the preset requirements;

[0121] Input the filter type, the frequency division point, and the filter order into the filter constructor to determine the filter coefficients;

[0122] The target filter is obtained based on the filtering coefficients.

[0123] In one embodiment, the filtering module 30 is further configured to use the spliced ​​segmented audio signal corresponding to each filtering channel as the input signal when the target filter is a finite impulse response filter;

[0124] The input signal is input into the finite impulse response filter to obtain the filtered segmented audio signal corresponding to each filtering channel.

[0125] In one embodiment, the splicing module 20 is further configured to determine the number of splicing sampling points based on the filter order;

[0126] Based on the number of splicing sampling points and the preset sampling positions, the splicing sampling points are determined in the initial segmented audio signals corresponding to each filtered channel.

[0127] In one embodiment, the filtering module 30 is further configured to acquire historical filtered audio signals corresponding to each filtering channel when the target filter is an infinite impulse response filter;

[0128] The spliced ​​segmented audio signal corresponding to each filtering channel is used as the first input signal, and the historical filtered audio signal corresponding to each filtering channel is used as the second input signal.

[0129] The first input signal and the second input signal are input into the infinite impulse response filter to obtain the filtered segmented audio signal.

[0130] In one embodiment, the filter order includes a first filter order and a second filter order, and the splicing module 20 is further configured to determine the number of splicing sampling points based on the first filter order;

[0131] Based on the number of splicing sampling points and the preset sampling positions, the splicing sampling points are determined in the initial segmented audio signals corresponding to each filtered channel;

[0132] The number of historical filtered audio signal sampling points is determined based on the second filter order;

[0133] Based on the number of sampling points of the historical filtered audio signal, the historical filtered audio signal corresponding to each filtered channel is determined.

[0134] In one embodiment, the filtering module 30 is further configured to use the current segmented audio signal as a new initial segmented audio signal, determine and retain the splicing sampling points corresponding to each filtering channel based on the new initial segmented audio signal, and return to the step of reading the current segmented audio signal received by each filtering channel.

[0135] It should be understood that the above are merely illustrative examples and do not constitute any limitation on the technical solution of the present invention. In specific applications, those skilled in the art can make settings as needed, and the present invention does not impose any restrictions on this.

[0136] It should be noted that the workflow described above is merely illustrative and does not limit the scope of protection of this invention. In practical applications, those skilled in the art can select some or all of the workflow to achieve the purpose of this embodiment according to actual needs, and no restrictions are imposed here.

[0137] In addition, for technical details not described in detail in this embodiment, please refer to the segmented audio signal filtering method provided in any embodiment of the present invention, which will not be repeated here.

[0138] Furthermore, 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 system 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 system. Unless otherwise specified, an element defined by the phrase "comprising one..." does not exclude the presence of other identical elements in the process, method, article, or system that includes that element.

[0139] The sequence numbers of the embodiments of the present invention are for descriptive purposes only and do not represent the superiority or inferiority of the embodiments.

[0140] Through the above description of the embodiments, those skilled in the art can clearly understand that the methods of the 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 the present invention, or the part that contributes to the prior art, can be embodied in the form of a software product. This computer software product is stored in a storage medium (such as read-only memory (ROM) / RAM, magnetic disk, optical disk) and includes several instructions to cause a terminal device (which may be a mobile phone, computer, server, or network device, etc.) to execute the methods described in the various embodiments of the present invention.

[0141] The above are merely preferred embodiments of the present invention and do not limit the scope of the patent. Any equivalent structural or procedural transformations made based on the description and drawings of the present invention, or direct or indirect applications in other related technical fields, are similarly included within the scope of patent protection of the present invention.

Claims

1. A filtering method for segmented audio signals, characterized in that, The filtering method for the segmented audio signal includes: Construct the target filter according to the preset requirements; Read the current segmented audio signal received from each filter channel; Obtain the splicing sampling points of the initial segmented audio signal corresponding to each filter channel, and splice the splicing sampling points of each filter channel with the current segmented audio signal to obtain the spliced ​​segmented audio signal corresponding to each filter channel, wherein the initial segmented audio signal is the previous segmented audio signal; Based on the spliced ​​segmented audio signals corresponding to each filtering channel and the target filter, the filtered segmented audio signals corresponding to each filtering channel are obtained, and the length of the filtered segmented audio signals is equal to the length of the current segmented audio signal; The filtered segmented audio signals corresponding to each filtered channel are superimposed to obtain the target segmented audio signal, and the target segmented audio signal is input into the target channel. The step of obtaining the filtered segmented audio signal corresponding to each filtering channel based on the spliced ​​segmented audio signal corresponding to each filtering channel and the target filter includes: when the target filter is an infinite impulse response filter, acquiring the historical filtered audio signal corresponding to each filtering channel; using the spliced ​​segmented audio signal corresponding to each filtering channel as a first input signal and the historical filtered audio signal corresponding to each filtering channel as a second input signal; inputting the first input signal and the second input signal into the infinite impulse response filter to obtain the filtered segmented audio signal; The infinite impulse response filter has a first filter order and a second filter order. Before reading the current segmented audio signal received by each filter channel, the process further includes: determining the number of splicing sampling points based on the first filter order; determining the splicing sampling points in the initial segmented audio signal corresponding to each filter channel based on the number of splicing sampling points and a preset sampling position; determining the number of historical filtered audio signal sampling points based on the second filter order; and determining the historical filtered audio signal corresponding to each filter channel based on the number of historical filtered audio signal sampling points.

2. The method as described in claim 1, characterized in that, The step of constructing the target filter according to preset requirements includes: Based on the preset requirements, determine the filter type, crossover point, filter order, and filter constructor. Input the filter type, the frequency division point, and the filter order into the filter constructor to determine the filter coefficients; The target filter is obtained based on the filtering coefficients, and the target filter is either a finite impulse response filter or an infinite impulse response filter.

3. The method as described in claim 2, characterized in that, The step of obtaining the filtered segmented audio signal corresponding to each filtered channel based on the spliced ​​segmented audio signal corresponding to each filtered channel and the target filter includes: When the target filter is a finite impulse response filter, the spliced ​​segmented audio signal corresponding to each filter channel is used as the input signal; The input signal is input into the finite impulse response filter to obtain the filtered segmented audio signal corresponding to each filtering channel.

4. The method as described in claim 3, characterized in that, Before reading the currently segmented audio signal received from each filter channel, the process also includes: The number of splicing sampling points is determined based on the filter order; Based on the number of splicing sampling points and the preset sampling positions, the splicing sampling points are determined in the initial segmented audio signals corresponding to each filtered channel.

5. The method as described in claim 1 or 4, characterized in that, The step of superimposing the filtered segmented audio signals corresponding to each filtered channel to obtain the target segmented audio signal, and then inputting the target segmented audio signal into the target channel, further includes: The current segmented audio signal is used as the new initial segmented audio signal. Based on the new initial segmented audio signal, the splicing sampling points corresponding to each filter channel are determined and retained. Then, the process returns to the step of reading the current segmented audio signal received by each filter channel.

6. A filtering device for segmented audio signals, characterized in that, The filtering device for the segmented audio signal includes: The construction module is used to construct the target filter based on the preset filter type, preset frequency division point, and preset filter order. The splicing module is used to read the currently segmented audio signal received from each filter channel; The splicing module is also used to obtain the splicing sampling points of the initial segmented audio signal corresponding to each filter channel, and splice the splicing sampling points of each filter channel with the current segmented audio signal to obtain the spliced ​​segmented audio signal corresponding to each filter channel, wherein the initial segmented audio signal is the previous segmented audio signal; The filtering module is used to obtain the filtered segmented audio signal corresponding to each filtering channel based on the spliced ​​segmented audio signal corresponding to each filtering channel and the target filter, wherein the length of the filtered segmented audio signal is equal to the length of the current segmented audio signal; The filtering module is also used to superimpose the filtered segmented audio signals corresponding to each filtering channel to obtain the target segmented audio signal, and input the target segmented audio signal into the target channel; The filtering module is also used to acquire historical filtered audio signals corresponding to each filtering channel when the target filter is an infinite impulse response filter; The spliced ​​segmented audio signal corresponding to each filtering channel is used as the first input signal, and the historical filtered audio signal corresponding to each filtering channel is used as the second input signal. The first input signal and the second input signal are input into the infinite impulse response filter to obtain the filtered segmented audio signal; The infinite impulse response filter has a first filter order and a second filter order. The filtering module is also used to determine the number of splicing sampling points based on the first filter order. Based on the number of splicing sampling points and the preset sampling positions, the splicing sampling points are determined in the initial segmented audio signals corresponding to each filtered channel; The number of historical filtered audio signal sampling points is determined based on the second filter order; Based on the number of sampling points of the historical filtered audio signal, the historical filtered audio signal corresponding to each filtered channel is determined.

7. A filtering device for segmented audio signals, characterized in that, The device includes: a memory, a processor, and a segmented audio signal filtering program stored in the memory and executable on the processor, the segmented audio signal filtering program being configured to implement the steps of the segmented audio signal filtering method as described in any one of claims 1 to 5.

8. A storage medium, characterized in that, The storage medium stores a filtering program for segmented audio signals, which, when executed by a processor, implements the steps of the filtering method for segmented audio signals as described in any one of claims 1 to 5.

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