CN117040487B - Filtering method, device, equipment and storage medium for audio signal processing - Google Patents
Filtering method, device, equipment and storage medium for audio signal processing Download PDFInfo
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Abstract
The invention discloses a filtering method, device, equipment and storage medium for audio signal processing. The invention sets a plurality of sampling points for each frame of audio signal by carrying out framing treatment on the audio file, so as to obtain a framed audio signal; determining a plurality of sampling points of the filtered input signal corresponding to each sampling point of the filtered output signal according to the subscript of each sampling point of the framing audio signal to obtain the filtered input signal; the method comprises the steps of inputting a filtering input signal into an audio filter for filtering to obtain a filtering output signal, processing two adjacent discrete framing signals generated in digital filtering to enable the discrete signals to have continuity, avoiding signal delay after discretizing the signals, determining a plurality of sampling points of the filtering input signal corresponding to each sampling point of the filtering output signal through the subscript of each sampling point of the framing audio signal to obtain a processed signal, filtering the processed audio signal, and enabling audio to keep the audio listening quality after filtering under the condition of a digital filtering mode.
Description
Technical Field
The present invention relates to the field of audio signal processing technologies, and in particular, to a filtering method, apparatus, device, and storage medium for audio signal processing.
Background
In the real world, the number of channels of the audio file is usually 2, and in order to obtain a more realistic feeling, a 7.1.4 panoramic sound system is usually adopted to display more music details, and the 7.1.4 panoramic sound system is composed of a total 12-channel system composed of 7 front-back left-right surrounding speakers, 1 subwoofer and 4 sky speakers, and two-channel stereo conversion into 7.1.4 panoramic sound is required when the 2-channel audio file is played by using 12 channels. When the two-channel stereo is converted into 7.1.4 panoramic sound, filtering is carried out on the split-frame audio signals, and when filtering is carried out, if filtering is carried out by adopting a digital method, each frame of signal has delay caused by filtering, so that each frame of signal is discontinuous, and poor listening effect is caused.
The foregoing is provided merely for the purpose of facilitating understanding of the technical solutions of the present invention and is not intended to represent an admission that the foregoing is prior art.
Disclosure of Invention
The invention mainly aims to provide a filtering method, device, equipment and storage medium for audio signal processing, and aims to solve the technical problem that the hearing effect of a filtered signal is poor in the prior art in a digital filtering mode.
To achieve the above object, the present invention provides a filtering method of audio signal processing, the method comprising the steps of:
carrying out framing treatment on the audio file, and setting a plurality of sampling points for each frame of audio signal to obtain a framing audio signal;
determining a plurality of sampling points of the filtered input signal corresponding to each sampling point of the filtered output signal according to the subscript of each sampling point of the framing audio signal to obtain the filtered input signal;
and inputting the filtered input signal to an audio filter for filtering to obtain a filtered output signal.
Optionally, determining a plurality of sampling points of the filtered input signal corresponding to each sampling point of the filtered output signal according to the subscript of each sampling point of the framed audio signal to obtain the filtered input signal includes:
obtaining a sub-frame audio signal index according to the position of the sub-frame audio signal;
determining the position of the sampling point of the framing audio signal in the whole section of audio signal according to the sampling point index of the framing audio signal and the length of the framing audio signal to obtain the sampling point index of the whole section of audio signal;
and obtaining a filtered input signal according to the index of the sampling point of the whole section of audio signal, the filter order, the index of the filter coefficient and the length of the framing audio signal.
Optionally, the inputting the filtered input signal to an audio filter, to obtain a filtered output signal, includes:
and obtaining a filtered output signal according to the filtered input signal and the filter coefficient.
Optionally, before the filtering input signal is input to the audio filter to obtain the filtering output signal, the method further includes:
determining a filter frequency domain transfer function according to the filter type;
performing inverse Fourier transform on the frequency domain transfer function to obtain a time domain impulse response;
sampling the time domain impulse response according to a preset sampling frequency to obtain a discrete impulse response;
windowing the discrete impulse response to obtain a filter coefficient;
updating an audio filter according to the filter coefficients and the length of the framed audio signal.
Optionally, before updating the audio filter according to the filter coefficients and the length of the framed audio signal, the method further comprises:
determining the pointer variable type of the filter coefficient;
generating a finite impulse response filter according to a window function method to obtain the filter coefficient;
and updating the configuration file of the audio filter by the pointer variable type of the filter coefficient.
Optionally, determining a plurality of sampling points of the filtered input signal corresponding to each sampling point of the filtered output signal according to the subscript of each sampling point of the framed audio signal, and before obtaining the filtered input signal, further includes:
determining pointer variable types of the filtered input signal and the filtered output signal, wherein the pointer variable types of the filtered input signal and the filtered output signal are the same;
and allocating corresponding memory spaces for the filtered input signals and the filtered output signals, wherein the lengths of the memory spaces corresponding to the filtered input signals and the filtered output signals are the same, and the memory spaces are used for storing the corresponding filtered input signals and the corresponding filtered output signals.
Optionally, the inputting the filtered input signal to an audio response filter to obtain a filtered output signal includes:
recording the current framing audio signal subscript;
and when the current sub-frame audio signal subscript is the last sub-frame audio signal of the audio file, obtaining a whole section of filtering signal according to the filtering output signal.
In addition, to achieve the above object, the present invention also proposes a filtering apparatus for audio signal processing, the filtering apparatus for audio signal processing including:
The audio framing module is used for framing the audio file, and setting a plurality of sampling points for each frame of audio signal to obtain a framed audio signal;
the audio processing module is used for determining a plurality of sampling points of the filtering input signal corresponding to each sampling point of the filtering output signal according to the subscript of each sampling point of the framing audio signal to obtain the filtering input signal;
and the audio filtering module is used for inputting the filtering input signal to an audio filter for filtering to obtain a filtering output signal.
In addition, to achieve the above object, the present invention also proposes a filtering apparatus for audio signal processing, the filtering apparatus for audio signal processing including: the audio signal processing device comprises a memory, a processor and an audio signal processing filter program stored on the memory and executable on the processor, wherein the audio signal processing filter program is configured to realize the steps of the audio signal processing filter method.
In addition, in order to achieve the above object, the present invention also proposes a storage medium having stored thereon a filter program for audio signal processing, which when executed by a processor, implements the steps of the filter method for audio signal processing as described above.
The invention sets a plurality of sampling points for each frame of audio signal by carrying out framing treatment on the audio file, so as to obtain a framed audio signal; determining a plurality of sampling points of the filtered input signal corresponding to each sampling point of the filtered output signal according to the subscript of each sampling point of the framing audio signal to obtain the filtered input signal; the filtering input signal is input into an audio filter for filtering to obtain a filtering output signal, the filtering output signal is processed according to two adjacent discrete framing signals generated in digital filtering, so that the discrete signals have continuity, signal delay caused by discretization of the signals is avoided, a plurality of sampling points of the filtering input signal corresponding to each sampling point of the filtering output signal are determined through the subscript of each sampling point of the framing audio signal, the processed signal is obtained, and the processed audio signal is filtered, so that the audio can keep the audio listening quality after filtering under the condition of a digital filtering mode.
Drawings
Fig. 1 is a schematic structural diagram of a filtering apparatus for audio signal processing in a hardware operating environment according to an embodiment of the present invention;
FIG. 2 is a flowchart of a filtering method for audio signal processing according to a first embodiment of the present invention;
FIG. 3 is a diagram illustrating a filtered input signal according to an embodiment of a filtering method for audio signal processing according to the present invention;
FIG. 4 is a flowchart illustrating a filtering method for audio signal processing according to a second embodiment of the present invention;
fig. 5 is a block diagram of a first embodiment of a filtering apparatus for audio signal processing according to the present invention.
The achievement of the objects, functional features and advantages of the present invention will be further described with reference to the accompanying drawings, in conjunction with the embodiments.
Detailed Description
It should be understood that the specific embodiments described herein are for purposes of illustration only and are not intended to limit the scope of the invention.
Referring to fig. 1, fig. 1 is a schematic diagram of a filtering device for audio signal processing in a hardware running environment according to an embodiment of the present invention.
As shown in fig. 1, the filtering apparatus of audio signal processing may include: a processor 1001, such as a central processing unit (Central Processing Unit, CPU), a communication bus 1002, a user interface 1003, a network interface 1004, a memory 1005. Wherein the communication bus 1002 is used to enable connected communication between these components. The user interface 1003 may include a Display, an input unit such as a Keyboard (Keyboard), and the optional user interface 1003 may further include a standard wired interface, a wireless interface. The network interface 1004 may optionally include a standard wired interface, a Wireless interface (e.g., a Wireless-Fidelity (Wi-Fi) interface). The Memory 1005 may be a high-speed random access Memory (Random Access Memory, RAM) Memory or a stable nonvolatile Memory (NVM), such as a disk Memory. The memory 1005 may also optionally be a storage device separate from the processor 1001 described above.
It will be appreciated by those skilled in the art that the structure shown in fig. 1 does not constitute a limitation of the filtering device of the audio signal processing, and may comprise more or less components than shown, or may combine certain components, or may be arranged in different components.
As shown in fig. 1, an operating system, a network communication module, a user interface module, and a filter program for audio signal processing may be included in the memory 1005 as one type of storage medium.
In the filtering apparatus for audio signal processing shown in fig. 1, the network interface 1004 is mainly used for data communication with a network server; the user interface 1003 is mainly used for data interaction with a user; the processor 1001 and the memory 1005 in the audio signal processing filtering apparatus of the present invention may be disposed in the audio signal processing filtering apparatus, and the audio signal processing filtering apparatus calls the audio signal processing filtering program stored in the memory 1005 through the processor 1001 and executes the audio signal processing filtering method provided in the embodiment of the present invention.
An embodiment of the present invention provides a filtering method for audio signal processing, and referring to fig. 2, fig. 2 is a flowchart of a first embodiment of a filtering method for audio signal processing according to the present invention.
In this embodiment, the filtering method for audio signal processing includes the following steps:
step S10: and carrying out framing processing on the audio file, and setting a plurality of sampling points for each frame of audio signal to obtain a framing audio signal.
It should be noted that, the execution body of the embodiment is a filtering device for audio signal processing, where the filtering device for audio signal processing has functions of data processing, data communication, program running, etc., and the filtering device for audio signal processing may be an integrated controller, a control computer, etc., or may be other devices with similar functions, which is not limited in this embodiment.
It should be understood that the 7.1.4 panoramic sound system is a total 12-channel system consisting of 7 front-back left-right surround speakers, 1 subwoofer, and 4 sky speakers, which presents more sound details through more speakers, creating a more realistic sound realistic sensation. In the real world, the number of channels of an audio file is typically 2, and in order to play a 2-channel audio file using 12 channels for better listening experience, a two-channel stereo to 7.1.4 panoramic sound is required. The left and right channels of the two-channel stereo are marked as L and R,7 front, back, left and right surrounding speakers in the 12 channels of the 7.1.4 panoramic sound are marked as FL, FC, FR, SL, SR, BL and BR,1 subwoofer is marked as LFE, and 4 sky speakers are marked as TFL, TFR, TBL and TBR. In the process of up-sampling a signal of converting a dual-channel stereo into 7.1.4 panoramic sound, L is subjected to high-pass filtering and then outputted by FL, R is subjected to high-pass filtering and then outputted by FR, L and R are added first and then outputted by FC after high-pass filtering, L and R are added first and then outputted by LFE after low-pass filtering, L is subjected to band-pass filtering and high-pass filtering and outputted by SL, BL, TFL and TBL, and R is subjected to band-pass filtering and high-pass filtering and outputted by SR, BR, TFR and TBR; LFE outputs low-frequency signals below 120Hz, signals output by other 11 channels are above 120Hz, SL, SR, BL and BR output medium-low frequency signals of 120Hz-2500Hz, TFL, TFR, TBL and TBR output medium-high frequency signals of above 2500 Hz; in addition, a certain gain adjustment and delay adjustment are required to be performed on signals output by 12 channels. If the off-line mode is used for filtering the audio signal, firstly, the whole section of audio is read locally, then the whole section of audio signal is filtered, the filtered whole section of audio signal is stored, and finally, the whole section of audio is read and output by a loudspeaker after the filtering. After filtering, the signal is continuous, although there is some delay in the signal, without affecting the listening effect. The method is realized in the realization of an embedded platform for converting the dual-channel stereo sound into 7.1.4 panoramic sound, the embedded platform can realize digital method filtering and analog method filtering, and when the digital method filtering is used, delay caused by the filtering exists between each frame of signals, so that the signals of each frame are discontinuous, and the listening effect is poor; if the filtering is not implemented digitally at the time, but rather an analog method of multipath fading recombination is used, a loss of accuracy in the signal filtering will result.
It can be understood that the audio file is audio that needs to be played currently, when the audio file is played, the audio file can be pre-stored in the audio signal processing filtering device, the audio signal processing filtering device can frame the audio file, and a plurality of sampling points are added to each frame of signal, the frame division mode is related to the attribute of the filtering device, the purpose of the frame division is to enable unsteady audio to be in a steady state in short time, the sampling points are related to the sampling frequency of the audio, the common sampling frequencies are five levels of 110 hz,22050hz,24000hz,44 hz,48000hz, and the embodiment does not limit the sampling frequency, the sampling points and the like.
In a specific implementation, the filtering device for audio signal processing can process an audio signal, frame the audio signal, and enable a long section of audio to be cut into a plurality of small segments, and for the whole section of audio file, the longer the length, the worse the steady state, so that in order to enable the audio filtering effect to be better, the frame the audio file can be processed, the whole section of audio file can be converted into a plurality of audio frames, and in order to facilitate the subsequent filtering operation, the length of each audio frame should be kept consistent, and all be set to a plurality of sampling points, and the 1024 sampling points are used for subsequent description in this embodiment, which is not limited by this embodiment.
Step S20: and determining a plurality of sampling points of the filtered input signal corresponding to each sampling point of the filtered output signal according to the index of each sampling point of the framing audio signal to obtain the filtered input signal.
It should be noted that, after framing the audio file, each audio frame can be labeled to describe the time sequence and the position sequence between frames, and for a selected one of the framed audio signals, the one framed audio signal preceding the current framed audio signal is the last framed audio signal, and each framed audio signal contains a plurality of sampling points with the same number. The filtering input signal refers to an audio signal input to a filtering device for audio signal processing, and a plurality of sampling points of the filtering input signal corresponding to each sampling point of the filtering output signal are determined through the subscript of each sampling point of the framing audio signal, so as to obtain the filtering input signal.
In a specific implementation, referring to fig. 3, fig. 3 is a schematic diagram of a filtered input signal. Since the audio file is currently processed in such a manner that the audio file is sampled according to a preset sampling frequency, a fixed number of sampling points are combined into one frame of audio signal, for example, every 1024 sampling points are used as one frame of audio signal, for two adjacent frame of audio signals, there is no correlation between two frame of audio signals, and there is a relatively independent relationship between two frame of audio signals, for example, the reference numeral 1023 (the subscript starts from 0) of the last sampling point of the previous frame, then the first sampling point of the following frame is 1024, and from the viewpoint of the frame to frame, there is no correlation between two frames, therefore, when filtering is performed, the filtering is performed on different audio signals, and therefore, the filtering effect is not expected. If during filtering, a plurality of sampling points of a filtering input signal corresponding to the current sampling point of the filtering output signal are determined according to the subscript of the current sampling point of the current framing audio signal, each sampling point of the filtering output signal corresponds to a plurality of sampling points of the filtering input signal, each sampling point of the current framing audio signal is traversed in sequence, and a plurality of sampling points of the filtering input signal corresponding to each sampling point of the filtering output signal are determined, so that the filtering input signal is obtained, continuity is presented between 2048 sampling points from the sampling point with the subscript of 0 to the sampling point with the subscript of 2047, continuity of filtering effects of two sections of framing audio signals can be guaranteed, and when the filtering input signal is stored, the filtering input signal can be stored through a floating point array.
Further, determining a plurality of sampling points of the filtered input signal corresponding to each sampling point of the filtered output signal according to the subscript of each sampling point of the framed audio signal to obtain the filtered input signal, including:
obtaining a sub-frame audio signal index according to the position of the sub-frame audio signal;
determining the position of the sampling point of the framing audio signal in the whole section of audio signal according to the sampling point index of the framing audio signal and the length of the framing audio signal to obtain the sampling point index of the whole section of audio signal;
and obtaining a filtered input signal according to the index of the sampling point of the whole section of audio signal, the filter order, the index of the filter coefficient and the length of the framing audio signal.
It should be noted that, the frame audio signal subscript, the signal sampling point subscript of the frame audio signal, and the whole audio signal sampling point subscript are all the position information for describing the current audio signal, and the position and the signal composition of the current signal can be located according to the subscript information.
In a specific implementation, since a plurality of framed audio signals are obtained after framing an audio file, each framed audio signal is marked with a subscript for describing a position where a current framed audio signal is located, for example, the subscript of a first framed audio signal may be set to 0, and an nth framed audio signal may be set to n-1. Similarly, when processing sampling points in a framed audio signal, the sampling points of the audio signal can be labeled in a similar manner, and if each of the current framed audio signals contains 1024 sampling points, the sampling points in the framed audio signal can be labeled 0000,0001. The embodiment does not limit the labeling format of the subscripts.
Step S30: and inputting the filtered input signal to an audio filter for filtering to obtain a filtered output signal.
It should be noted that the filtered output signal refers to an output signal obtained after filtering by an audio filter, and has a better listening effect compared with the filtered input signal, and meanwhile, the output length of the filtered output signal is the signal length corresponding to the current frame, and the filtered output signal is stored in a floating point array with the length consistent with the length of the current frame.
Further, the inputting the filtered input signal to an audio filter to obtain a filtered output signal includes:
and obtaining a filtered output signal according to the filtered input signal and the filter coefficient.
It should be noted that, in the implementation of the embedded platform for converting the dual-channel stereo sound into the 7.1.4 panoramic sound, a digital method is used to filter the framing audio signal with the length of L, according to the principle of signal filtering, if the input is a signal with the length of (n+l) formed by splicing the last N sampling points of the previous frame audio signal and the current frame audio signal with the length of L, then the audio signal with the length of L output after filtering does not have a bad listening effect caused by framing filtering, so that the filtering result of the framing audio signal is consistent with the filtering result of the whole section of audio signal, but at this time, the input of the filtering function is a floating point array with the length of (n+l), and the output is a floating point array with the length of L, so that more memory needs to be allocated for the input. When filtering, filtering is carried out according to a finite impulse response filter, and a specific filtering formula is as follows:
（1）
In equation (1), y is the output signal, x is the input signal, a is the filter coefficient, N is the filter order, N is the signal sample point index, and k is the filter coefficient index. As is available from the filtering formula, the current output signal y (N) is a weighted sum of the current input signal x (N) and the previous N input signals. Finally, a filtered output signal y (n) is obtained.
In a specific implementation, in an embedded platform implementation of converting a dual-channel stereo sound into a 7.1.4 panoramic sound, a digital method is used for filtering a framing audio signal with a length of L, according to a principle of signal filtering, if a signal with a length of N+L, which is formed by splicing the last N sampling points of a previous frame audio signal and a current frame audio signal with a length of L, is input, then the audio signal with the length of L, which is output after filtering, does not have a bad listening effect caused by framing filtering, so that a result of filtering the framing audio signal is consistent with a result of filtering the whole section of audio signal, but at the moment, when the input length of a filtering function is the floating point array with the length of N+L, a floating point array with the length of L is output, and more memory needs to be allocated for the input, so that the memory required by filtering is increased, therefore, the memory allocation of the input and the output can be optimized on the memory through a formula 2 and a formula 3, and the memory space is saved, and the formulas 2 and 3 are respectively:
（2）
（3）
Where L is the length of each frame of signal, i is the index of the number of frames of signal, j is the index of the sampling point of each frame of signal, N is the index of the sampling point of the whole section of signal, y is the output signal, x is the input signal, N is the filter order, a is the filter coefficient, and k is the index of the filter coefficient. Equation 2 finds the subscript of the current sampling point of the current frame signal that is located in the whole signal. Equation 3 adds a modulo operation (%) to equation 1. The invention constructs a filter function according to a formula 2 and a formula 3, wherein the input parameters of the function are x, a, i, j, N and L, the output parameters of the function are y, x and a are floating point type pointer variables, i, j, N and L are integer type variables, and y is a floating point type variable. The signal length L per frame typically takes a value of 1024. The larger the value of the filter order N, the higher the filter accuracy, but the larger the calculation amount, the larger the delay, and the filter order N is usually set to several hundred, for example, 200, in consideration of compromise, which is not limited to this in the present embodiment.
Referring to fig. 3, for the 1 st frame input data and the 2 nd frame input data, according to formula (1), in order to keep continuous after the frame audio signal is filtered, it is necessary to splice the tail of the 1 st frame input data and the 2 nd frame input data as the filtered input signal, then to inverse-fold the filter coefficient, align the filtered input signal with the filtered input signal, multiply the filtered input signal sampling point by the filter coefficient correspondingly, then accumulate, obtain the filtered output sampling point, shift the filter coefficient rightwards, multiply by the filtered input signal sampling point correspondingly, then accumulate, sequentially obtain the filtered output sampling point, and obtain the filtered output signal. The method comprises the steps of firstly splicing the tail part of the previous frame with the current frame signal to obtain a complete filtering input signal, and then performing filtering calculation, so that the length of the filtering input signal is N+L (N is the filter order, L is the frame-divided audio signal length), and the length of the filtering output signal is L. According to the formula (2) and the formula (3), the filtering calculation is performed while the audio signal of the current frame is read, so that the length of the filtering input signal and the length of the filtering output signal are L, when the 1 st sampling point of the filtering output signal is calculated (the subscript starts from 0), the 7 st, 8, 9 and 10 sampling points of the 1 st frame in the filtering input signal are needed to be known, and when the audio signal of the current frame is read while the filtering calculation is performed, the 1 st sampling point of the audio signal of the current frame (the 2 nd frame) is calculated, the 1 st sampling point of the audio signal of the current frame is only read and stored at the 0 position of the subscript of the filtering input signal, the 7 st, 8 th and 10 th sampling points of the 1 st frame are still stored at the 6 th, 7, 8 th and 9 th sampling points of the filtering input signal, and the like, the filtering calculation is performed while the audio signal of the current frame is read, the filtering input signal is updated, and the lengths of the filtering input signal and the filtering output signal are L.
Further, the inputting the filtered input signal to an audio response filter to obtain a filtered output signal includes:
recording the current framing audio signal subscript;
and when the current sub-frame audio signal subscript is the last sub-frame audio signal of the audio file, obtaining a whole section of filtering signal according to the filtering output signal.
In a specific implementation, after filtering the current frame audio signal, the subscript of the current frame audio signal needs to be recorded, because the audio file length is known, the total frame number of the frame audio signal of the audio file is also known, so that the filtering progress can be known according to the subscript of the current frame audio signal, if the subscript of the current frame audio signal is not the last frame of frame audio signal, the next frame of the current frame is continuously filtered, and if the subscript of the current frame audio signal is equal to the total frame number, after the filtering of the current frame audio signal is completed, the filtering operation is stopped, and a whole-section filtering signal of a whole-section audio is obtained according to the filtering output signal.
In the embodiment, the frame-dividing processing is performed on the audio file, and a plurality of sampling points are set for each frame of audio signal to obtain a frame-dividing audio signal; determining a plurality of sampling points of the filtered input signal corresponding to each sampling point of the filtered output signal according to the subscript of each sampling point of the framing audio signal to obtain the filtered input signal; the filtering input signal is input into an audio filter for filtering to obtain a filtering output signal, the filtering output signal is processed according to two adjacent discrete framing signals generated in digital filtering, so that the discrete signals have continuity, signal delay caused by discretization of the signals is avoided, a plurality of sampling points of the filtering input signal corresponding to each sampling point of the filtering output signal are determined through the subscript of each sampling point of the framing audio signal, the processed signal is obtained, and the processed audio signal is filtered, so that the audio can keep the audio listening quality after filtering under the condition of a digital filtering mode.
Referring to fig. 4, fig. 4 is a flowchart illustrating a filtering method for audio signal processing according to a second embodiment of the present invention.
Based on the first embodiment, the filtering method for audio signal processing of the present embodiment further includes, before the step S20:
step S201: obtaining a sub-frame audio signal index according to the position of the sub-frame audio signal;
step S202: determining the position of the sampling point of the framing audio signal in the whole section of audio signal according to the sampling point index of the framing audio signal and the length of the framing audio signal to obtain the sampling point index of the whole section of audio signal;
step S203: and obtaining a filtered input signal according to the index of the sampling point of the whole section of audio signal, the filter order, the index of the filter coefficient and the length of the framing audio signal.
In the specific implementation, firstly, determining a main body framing audio signal which is currently filtered, wherein the main body framing audio signal is a currently selected frame, determining the position of the current framing audio signal, and determining the subscript of the framing audio signal according to the position; after the current sub-frame audio signal subscript is obtained, the subscript of the signal sampling point of the current sub-frame audio signal can be determined; determining the position of the current sampling point in the whole section of audio signal according to the current sampling point index of the current framing audio signal and the current framing audio signal index and combining the length of the framing audio signal to obtain the whole section of audio signal sampling point index; each sampling point of the filtering output signal corresponds to a plurality of sampling points of the filtering input signal, the number of the sampling points of the filtering input signal is equal to the number of filter coefficients and is also equal to the number of filter orders plus one, the index of the whole section of audio signal minus the index of the filter coefficients (the index value starts from 0 to the end of the filter orders) is subjected to modulo division on the length of the audio signal, the sampling points of the filtering input signal corresponding to the current sampling point of the filtering output signal are determined, each sampling point of the current frame-divided audio signal is traversed in sequence, and the sampling points of the filtering input signal corresponding to each sampling point of the filtering output signal are determined, so that the filtering input signal is obtained.
In framing an audio file, a certain number of sampling points may be combined into one frame of an audio signal, and thus, there are a plurality of sampling points for one frame of an audio signal. The filtering of the audio signal processing can identify each frame of audio signal, determine the number of sampling points in each frame of audio signal, and label the sampling points to obtain subscripts of the sampling points.
Further, before inputting the filtered input signal to the audio filter to obtain the filtered output signal, the method further includes:
determining a filter frequency domain transfer function according to the filter type;
performing inverse Fourier transform on the frequency domain transfer function to obtain a time domain impulse response;
sampling the time domain impulse response according to a preset sampling frequency to obtain a discrete impulse response;
windowing the discrete impulse response to obtain a filter coefficient;
updating an audio filter according to the filter coefficients and the length of the framed audio signal.
It should be noted that, the audio filter is one of the components in the filtering device for audio signal processing, and is capable of filtering an audio file, where the audio filter includes an audio filtering strategy.
In a specific implementation, a filter frequency domain transfer function is first determined according to a type of a filter, where the type of the filter may be high-pass, low-pass, band-pass or band-stop, and the type of the filter has a correspondence with a corresponding frequency domain transfer function, and the correspondence may be recorded in a type-function matching table and stored in a filtering device for processing the audio signal. After the frequency domain transfer function of the filter is determined according to the filter type, the inverse Fourier transform can be carried out on the transfer function to obtain a time domain impulse response, the impulse response is sampled according to a certain sampling frequency to be discretized, meanwhile, a proper window function is selected, the discrete impulse response is windowed to obtain a discrete impulse response with a finite length, and the coefficient of the filter is obtained.
Further, determining a plurality of sampling points of the filtered input signal corresponding to each sampling point of the filtered output signal according to the subscript of each sampling point of the framed audio signal, and before obtaining the filtered input signal, further includes:
Determining pointer variable types of the filtered input signal and the filtered output signal, wherein the pointer variable types of the filtered input signal and the filtered output signal are the same;
and allocating corresponding memory spaces for the filtered input signals and the filtered output signals, wherein the lengths of the memory spaces corresponding to the filtered input signals and the filtered output signals are the same, and the memory spaces are used for storing the corresponding filtered input signals and the corresponding filtered output signals.
Further, before updating the audio filter according to the filter coefficients and the length of the framed audio signal, the method further comprises:
determining the pointer variable type of the filter coefficient;
generating a finite impulse response filter according to a window function method to obtain the filter coefficient;
and updating the configuration file of the audio filter by the pointer variable type of the filter coefficient.
In a specific implementation, in the process of filtering a split-frame audio signal, traversing an input signal point by point, calculating an output signal point by point, and the program flow is as follows:
declaring the input and the output as floating point pointer variables, and distributing a memory with the length L for the input and the output;
The filter coefficients are stated as floating point pointer variables, a finite impulse response filter is designed by using a window function method, the filter coefficients are obtained, and the filter coefficients can be written into a configuration file for updating an audio filter for standardizing codes and facilitating modification.
Reading a framing audio signal with the length L, and assigning the read framing audio signal to an input variable;
traversing an input signal, recording a frame index and a sampling point index of a current frame signal, calling a filtering function, solving the index of the current sampling point of the current frame signal positioned on the whole section of signal according to a formula 2, and calculating an output signal according to a formula 3;
after the current frame signal is processed, the step 3 and the step 4 are cycled to process the next frame signal;
and after the last frame of signal is processed, the cycle is finished, and the whole section of signal filtering is completed.
The input signal of the constructed filter function is not a floating point type array with the length of (N+L), but a floating point type pointer with the length of L is allocated to the memory, and the pointer operation is faster than the array operation.
According to the embodiment, the input and output of the floating point array are converted into floating point pointer variables, memories with the same length are distributed for the input and the output, the situation that the memory required by the input is larger than the output for ensuring the filtering effect is solved by optimizing a filtered output formula, and the aim of improving the listening effect in a digital filtering mode is fulfilled under the condition that the input memory is equal to the output memory.
In addition, the embodiment of the invention also provides a storage medium, wherein the storage medium stores a filtering program of audio signal processing, and the filtering program of audio signal processing realizes the steps of the filtering method of audio signal processing when being executed by a processor.
Referring to fig. 5, fig. 5 is a block diagram illustrating a first embodiment of a filtering apparatus for audio signal processing according to the present invention.
As shown in fig. 5, a filtering apparatus for audio signal processing according to an embodiment of the present invention includes:
the audio framing module 10 is used for framing the audio file, and setting a plurality of sampling points for each frame of audio signal to obtain a framed audio signal;
the audio processing module 20 is configured to determine a plurality of sampling points of the filtered input signal corresponding to each sampling point of the filtered output signal according to the subscript of each sampling point of the framed audio signal, so as to obtain the filtered input signal;
the audio filtering module 30 is configured to input the filtered input signal to an audio filter for filtering, so as to obtain a filtered output signal.
In the embodiment, the frame-dividing processing is performed on the audio file, and a plurality of sampling points are set for each frame of audio signal to obtain a frame-dividing audio signal; determining a plurality of sampling points of the filtered input signal corresponding to each sampling point of the filtered output signal according to the subscript of each sampling point of the framing audio signal to obtain the filtered input signal; the filtering input signal is input into an audio filter for filtering to obtain a filtering output signal, the filtering output signal is processed according to two adjacent discrete framing signals generated in digital filtering, so that the discrete signals have continuity, signal delay caused by discretization of the signals is avoided, a plurality of sampling points of the filtering input signal corresponding to each sampling point of the filtering output signal are determined through the subscript of each sampling point of the framing audio signal, the processed signal is obtained, and the processed audio signal is filtered, so that the audio can keep the audio listening quality after filtering under the condition of a digital filtering mode.
In an embodiment, the audio processing module 20 is further configured to obtain a frame audio signal subscript according to a location of the frame audio signal; determining the position of the sampling point of the framing audio signal in the whole section of audio signal according to the sampling point index of the framing audio signal and the length of the framing audio signal to obtain the sampling point index of the whole section of audio signal; and obtaining a filtered input signal according to the index of the sampling point of the whole section of audio signal, the filter order, the index of the filter coefficient and the length of the framing audio signal.
In one embodiment, the audio filtering module 30 obtains a filtered output signal according to the filtered input signal and the filter coefficients.
In an embodiment, the audio processing module 20 is further configured to determine a filter frequency domain transfer function according to a filter type; performing inverse Fourier transform on the frequency domain transfer function to obtain a time domain impulse response; sampling the time domain impulse response according to a preset sampling frequency to obtain a discrete impulse response; windowing the discrete impulse response to obtain a filter coefficient; updating an audio filter according to the filter coefficients and the length of the framed audio signal.
In an embodiment, the audio processing module 20 is further configured to determine a pointer variable type of the filter coefficient; generating a finite impulse response filter according to a window function method to obtain the filter coefficient; and updating the configuration file of the audio filter by the pointer variable type of the filter coefficient.
In an embodiment, the audio processing module 20 is further configured to determine a pointer variable type of the filtered input signal and the filtered output signal, where the pointer variable type of the filtered input signal and the filtered output signal are the same; and allocating corresponding memory spaces for the filtered input signals and the filtered output signals, wherein the lengths of the memory spaces corresponding to the filtered input signals and the filtered output signals are the same, and the memory spaces are used for storing the corresponding filtered input signals and the corresponding filtered output signals.
In one embodiment, the audio filtering module 30 is further configured to record a current frame audio signal index; and when the current sub-frame audio signal subscript is the last sub-frame audio signal of the audio file, obtaining a whole section of filtering signal according to the filtering output signal.
It should be understood that the foregoing is illustrative only and is not limiting, and that in specific applications, those skilled in the art may set the invention as desired, and the invention is not limited thereto.
It should be noted that the above-described working procedure is merely illustrative, and does not limit the scope of the present invention, and in practical application, a person skilled in the art may select part or all of them according to actual needs to achieve the purpose of the embodiment, which is not limited herein.
Furthermore, it should be noted that, in this document, the terms "comprises," "comprising," or any other variation thereof, are intended to cover a non-exclusive inclusion, such that a process, method, article, or system that comprises a list of elements does not include only those elements but may include other elements not expressly listed or inherent to such process, method, article, or system. Without further limitation, an element defined by the phrase "comprising one … …" does not exclude the presence of other like elements in a process, method, article, or system that comprises the element.
The foregoing embodiment numbers of the present invention are merely for the purpose of description, and do not represent the advantages or disadvantages of the embodiments.
From the above description of embodiments, it will be clear to a person skilled in the art that the above embodiment method may be implemented by means of software plus a necessary general hardware platform, but may of course also be implemented by means of hardware, but in many cases the former is a preferred embodiment. Based on such understanding, the technical solution of the present invention may be embodied essentially or in a part contributing to the prior art in the form of a software product stored in a storage medium (e.g. Read Only Memory (ROM)/RAM, magnetic disk, optical disk) and comprising several instructions for causing a terminal device (which may be a mobile phone, a computer, a server, or a network device, etc.) to perform the method according to the embodiments of the present invention.
The foregoing description is only of the preferred embodiments of the present invention, and is not intended to limit the scope of the invention, but rather is intended to cover any equivalents of the structures or equivalent processes disclosed herein or in the alternative, which may be employed directly or indirectly in other related arts.
Claims (9)
1. A filtering method of audio signal processing, characterized in that the filtering method of audio signal processing comprises:
carrying out framing treatment on the audio file, and setting a plurality of sampling points for each frame of audio signal to obtain a framing audio signal;
each sampling point of the filtering output signal corresponds to a plurality of sampling points of the filtering input signal, and a framing audio signal index is obtained according to the position of the framing audio signal; determining the position of the sampling point of the framing audio signal in the whole section of audio signal according to the sampling point index of the framing audio signal and the length of the framing audio signal to obtain the sampling point index of the whole section of audio signal; according to the whole section of audio signal sampling point subscript, the filter order, the filter coefficient subscript and the length of the framing audio signal, a filtering input signal is obtained, and a calculation formula is as follows:
；/>
wherein, the memory allocation of the input and output is L, L is the length of each frame of signal, i is the index of the frame number of the signal, j is the index of the sampling point of each frame of signal, N is the index of the sampling point of the whole section of signal, y is the output signal, x is the input signal, N is the filter order, a is the filter coefficient, k is the index of the filter coefficient;
And inputting the filtered input signal to an audio filter for filtering to obtain a target filtered output signal.
2. The method of claim 1, wherein inputting the filtered input signal to an audio filter results in a target filtered output signal, comprising:
and obtaining the target filtering output signal according to the filtering input signal and the filter coefficient.
3. The method of claim 1, wherein before inputting the filtered input signal to an audio filter to obtain a target filtered output signal, further comprising:
determining a filter frequency domain transfer function according to the filter type;
performing inverse Fourier transform on the frequency domain transfer function to obtain a time domain impulse response;
sampling the time domain impulse response according to a preset sampling frequency to obtain a discrete impulse response;
windowing the discrete impulse response to obtain a filter coefficient;
updating an audio filter according to the filter coefficients and the length of the framed audio signal.
4. A method according to claim 3, wherein before updating the audio filter based on the filter coefficients and the length of the framed audio signal, further comprising:
Determining the pointer variable type of the filter coefficient;
generating a finite impulse response filter according to a window function method to obtain the filter coefficient;
and updating the configuration file of the audio filter by the pointer variable type of the filter coefficient.
5. The method of claim 1, wherein each sampling point of the filtered output signal corresponds to a number of sampling points of the filtered input signal, and wherein a framed audio signal index is obtained based on a location of the framed audio signal; determining the position of the sampling point of the framing audio signal in the whole section of audio signal according to the sampling point index of the framing audio signal and the length of the framing audio signal to obtain the sampling point index of the whole section of audio signal; according to the whole audio signal sampling point index, the filter order, the filter coefficient index and the length of the framing audio signal, before obtaining the filtering input signal, the method further comprises:
determining pointer variable types of the filtered input signal and the filtered output signal, wherein the pointer variable types of the filtered input signal and the filtered output signal are the same;
And allocating corresponding memory spaces for the filtered input signals and the filtered output signals, wherein the lengths of the memory spaces corresponding to the filtered input signals and the filtered output signals are the same, and the memory spaces are used for storing the corresponding filtered input signals and the corresponding filtered output signals.
6. The method of any one of claims 1 to 5, wherein said inputting the filtered input signal to an audio response filter to obtain a target filtered output signal comprises:
recording the current framing audio signal subscript;
and when the current sub-frame audio signal subscript is the last sub-frame audio signal of the audio file, obtaining a whole section of filtering signal according to the filtering output signal.
7. A filtering apparatus for audio signal processing, characterized in that the filtering apparatus for audio signal processing comprises:
the audio framing module is used for framing the audio file, and setting a plurality of sampling points for each frame of audio signal to obtain a framed audio signal;
the audio processing module is used for obtaining a frame audio signal index according to the position of the frame audio signal, wherein each sampling point of the filtering output signal corresponds to a plurality of sampling points of the filtering input signal; determining the position of the sampling point of the framing audio signal in the whole section of audio signal according to the sampling point index of the framing audio signal and the length of the framing audio signal to obtain the sampling point index of the whole section of audio signal; according to the whole section of audio signal sampling point subscript, the filter order, the filter coefficient subscript and the length of the framing audio signal, a filtering input signal is obtained, and a calculation formula is as follows:
；/>
Wherein, the memory allocation of the input and output is L, L is the length of each frame of signal, i is the index of the frame number of the signal, j is the index of the sampling point of each frame of signal, N is the index of the sampling point of the whole section of signal, y is the output signal, x is the input signal, N is the filter order, a is the filter coefficient, k is the index of the filter coefficient;
and the audio filtering module is used for inputting the filtering input signal to an audio filter for filtering to obtain a target filtering output signal.
8. A filtering apparatus for audio signal processing, the apparatus comprising: memory, a processor and an audio signal processing filter program stored on the memory and executable on the processor, the audio signal processing filter program being configured to implement the steps of the audio signal processing filter method according to any one of claims 1 to 6.
9. A storage medium, wherein a filter program for audio signal processing is stored on the storage medium, which when executed by a processor, implements the steps of the filter method for audio signal processing according to any one of claims 1 to 6.
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