CN107481727B - Audio signal processing method and system based on electric tone pitch control - Google Patents

Audio signal processing method and system based on electric tone pitch control Download PDF

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CN107481727B
CN107481727B CN201710485440.4A CN201710485440A CN107481727B CN 107481727 B CN107481727 B CN 107481727B CN 201710485440 A CN201710485440 A CN 201710485440A CN 107481727 B CN107481727 B CN 107481727B
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罗时志
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    • G10L21/00Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/003Changing voice quality, e.g. pitch or formants
    • G10L21/007Changing voice quality, e.g. pitch or formants characterised by the process used
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    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
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Abstract

The invention discloses an audio signal processing method and system based on electric tone pitch control, wherein the processing method comprises the following steps: the analog signal input end receives an input analog signal and sends the input analog signal to the audio codec from the analog signal input end; the audio codec converts an input analog signal received from an analog signal input end into an input digital signal and sends the input digital signal to the audio DSP; after receiving the input digital signal, the audio DSP identifies the frequency component in the input digital signal, adjusts the input digital signal according to the frequency component in the digital signal, an expected basic tone and a preset adjusting method to obtain an output digital signal, and sends the output digital signal to an audio codec; the audio codec converts an output digital signal received from the audio DSP into an output analog signal and sends the output analog signal to an analog signal output end; the analog signal output end outputs the received output analog signal.

Description

Audio signal processing method and system based on electric tone pitch control
Technical Field
The invention relates to the field of sound processing, in particular to an audio signal processing method and system based on electric tone pitch control.
Background
The popularity of electronic audio signals currently makes a wide demand for processing electronic audio signals, and among them, performing up-scaling, down-scaling, etc. on audio signals according to the required electrical tone pitch becomes an urgent demand in audio signal processing. Although some devices and software process audio signals, basically only the key of the audio signals can be calculated, and the audio signals cannot be adjusted according to the expected key. There is therefore a need for a method and system that can process audio signals according to the pitch of the electrical tones.
Disclosure of Invention
In order to solve the above problems, the present invention provides an audio signal processing method and system based on tone control.
The invention provides an audio signal processing method based on electric tone control, which is realized by an analog signal input end, an audio codec, an audio DSP (digital signal processor) and an analog signal output end, and comprises the following steps:
the analog signal input end receives an input analog signal and sends the input analog signal to the audio codec from the analog signal input end;
the audio codec converts an input analog signal received from an analog signal input end into an input digital signal and sends the input digital signal to the audio DSP;
after receiving the input digital signal, the audio DSP identifies the frequency component in the input digital signal, adjusts the input digital signal according to the frequency component in the digital signal, an expected basic tone and a preset adjusting method to obtain an output digital signal, and sends the output digital signal to an audio codec;
the audio codec converts an output digital signal received from the audio DSP into an output analog signal and sends the output analog signal to an analog signal output end;
the analog signal output end outputs the received output analog signal.
Preferably, the first and second liquid crystal materials are,
the audio DSP comprises a Digital Signal Processor (DSP),
a memory storing a processing program for performing spectrum analysis on the digital signal and processing the digital signal according to the parameter;
an electric tone base controller comprising parameters for electric tone base control, said parameters comprising 13 values, representing A, A #, B, C, C #, D, D #, E, F, F #, G, G #, and #13 electric tone base;
the MCU (MicroControllerUnit, namely a single chip microcomputer) comprises an initialization program and a control program and is used for initializing and controlling the audio codec and the audio DSP;
the MCU calls a processing program in the memory, processes the input digital signal, identifies frequency components in the input digital signal, and processes the input digital signal to perform fast Fourier transform on the input digital signal to obtain frequency domain information of the input digital signal; that is, the frequency domain information of the digital signal is obtained by a first formula, where the first formula is:
F(ω)=FFT(f(N))
wherein, F (ω) is the frequency domain information of the input digital signal, F (n) is the input digital signal, and FFT is fast fourier transform;
the MCU reads the parameters of the electric tone basic tone controller, and according to the frequency components in the input digital signals, the expected basic tone and a preset adjusting method, calls a processing program in the memory to adjust the input digital signals to obtain output digital signals, namely the output digital signals are obtained through a second formula, wherein the second formula is as follows:
S(N)=IFFT(G(kω)·F(kω))
wherein, F (k ω) is the frequency domain information F (ω) of the input digital signal, k is the parameter value for the control of the electrical tone, G (k ω) is the amplitude adjustment function, s (n) is the output digital signal, IFFT is the inverse fast fourier transform.
Preferably, the processing the input digital signal includes:
the MCU performs discrete Fourier transform on the input digital signal;
the MCU performs ideal filtering on the input digital signal after Fourier transform;
the MCU restores the filtered input digital signal to a time domain signal through inverse fourier transform.
Preferably, the MCU performs discrete fourier transform on the input digital signal, including:
the MCU performs windowing processing on the input digital signal, and the window function used for windowing is as follows:
Figure BDA0001330359050000031
where w (N) is the value of the window function, N is the frame length, and N is the sampling point.
Preferably, the MCU performs discrete fourier transform on the input digital signal, and may further implement:
the MCU performs short-time Fourier transform or wavelet transform on the input digital signal.
The invention also provides an audio signal processing system based on the electric tone pitch control, which consists of an analog signal input end, an audio codec, an audio DSP and an analog signal output end, and comprises the following components:
the analog signal input end receives an input analog signal and sends the input analog signal to the audio codec from the analog signal input end;
the audio codec converts an input analog signal received from an analog signal input end into an input digital signal and sends the input digital signal to the audio DSP;
after receiving the input digital signal, the audio DSP identifies the frequency component in the input digital signal, adjusts the input digital signal according to the frequency component in the digital signal, an expected basic tone and a preset adjusting method to obtain an output digital signal, and sends the output digital signal to an audio codec;
the audio codec converts an output digital signal received from the audio DSP into an output analog signal and sends the output analog signal to an analog signal output end;
the analog signal output end outputs the received output analog signal.
Preferably, the first and second liquid crystal materials are,
the audio DSP, including,
a memory storing a processing program for performing spectrum analysis on the digital signal and processing the digital signal according to the parameter;
an electric tone base controller comprising parameters for electric tone base control, said parameters comprising 13 values, representing A, A #, B, C, C #, D, D #, E, F, F #, G, G #, and #13 electric tone base;
the MCU comprises an initialization program and a control program and is used for initializing and controlling the audio coder-decoder and the audio DSP;
the MCU calls a processing program in the memory, processes the input digital signal, identifies frequency components in the input digital signal, and processes the input digital signal to perform fast Fourier transform on the input digital signal to obtain frequency domain information of the input digital signal; that is, the frequency domain information of the digital signal is obtained by a first formula, where the first formula is:
F(ω)=FFT(f(N))
wherein, F (ω) is the frequency domain information of the input digital signal, F (n) is the input digital signal, and FFT is fast fourier transform;
the MCU reads the parameters of the electric tone basic tone controller, and according to the frequency components in the input digital signals, the expected basic tone and a preset adjusting method, calls a processing program in the memory to adjust the input digital signals to obtain output digital signals, namely the output digital signals are obtained through a second formula, wherein the second formula is as follows:
S(N)=IFFT(G(kω)·F(kω))
wherein, F (k ω) is the frequency domain information F (ω) of the input digital signal, k is the parameter value for the control of the electrical tone, G (k ω) is the amplitude adjustment function, s (n) is the output digital signal, IFFT is the inverse fast fourier transform.
Preferably, the MCU processes the input digital signal, and includes:
the MCU performs discrete Fourier transform on the input digital signal;
the MCU performs ideal filtering on the input digital signal after Fourier transform;
the MCU restores the filtered input digital signal to a time domain signal through inverse fourier transform.
Preferably, the MCU performs discrete fourier transform on the input digital signal, and includes:
the MCU performs windowing processing on the input digital signal, and the window function used for windowing is as follows:
Figure BDA0001330359050000051
where w (N) is the value of the window function, N is the frame length, and N is the sampling point.
Preferably, the MCU performs discrete fourier transform on the input digital signal, and may further implement:
the MCU performs short-time Fourier transform or wavelet transform on the input digital signal.
Some of the benefits of the present invention may include:
the audio signal processing method and system based on the electric tone pitch control can process the audio signal according to the electric tone pitch, occupy small storage space and process the audio signal more accurately.
Additional features and advantages of the invention will be set forth in the description which follows, and in part will be obvious from the description, or may be learned by practice of the invention. The objectives and other advantages of the invention will be realized and attained by the structure particularly pointed out in the written description and claims hereof as well as the appended drawings.
The technical solution of the present invention is further described in detail by the accompanying drawings and embodiments.
Drawings
The accompanying drawings, which are included to provide a further understanding of the invention and are incorporated in and constitute a part of this specification, illustrate embodiments of the invention and together with the description serve to explain the principles of the invention and not to limit the invention. In the drawings:
FIG. 1 is a flowchart of an audio signal processing method based on electric tone control according to an embodiment of the present invention;
fig. 2 is a schematic diagram of an audio signal processing system based on electrical tone control according to an embodiment of the present invention.
Detailed Description
The preferred embodiments of the present invention will be described in conjunction with the accompanying drawings, and it will be understood that they are described herein for the purpose of illustration and explanation and not limitation.
Fig. 1 is a flowchart of an audio signal processing method based on electric tone control according to an embodiment of the present invention. As shown in fig. 1, the method is implemented by an analog signal input terminal, an audio codec, an audio DSP, and an analog signal output terminal, and includes:
s101, an analog signal input end receives an input analog signal and sends the input analog signal to an audio codec from the analog signal input end;
step S102, an audio codec converts an input analog signal received from an analog signal input end into an input digital signal and sends the input digital signal to an audio DSP;
step S103, after receiving the input digital signal, the audio DSP identifies the frequency component in the input digital signal, adjusts the input digital signal according to the frequency component in the digital signal, the expected tone and a preset adjusting method to obtain an output digital signal, and sends the output digital signal to an audio codec;
step S104, the audio coder-decoder converts the output digital signal received from the audio DSP into an output analog signal and sends the output analog signal to an analog signal output end;
and step S105, outputting the analog signal by the analog signal output end according to the received output analog signal.
According to the method provided by the invention, the audio signal can be processed according to the electric tone.
In one embodiment of the present invention,
the audio DSP, including,
a memory storing a processing program for performing spectrum analysis on the digital signal and processing the digital signal according to the parameter;
an electric tone base controller comprising parameters for electric tone base control, said parameters comprising 13 values, representing A, A #, B, C, C #, D, D #, E, F, F #, G, G #, and #13 electric tone base;
the MCU comprises an initialization program and a control program and is used for initializing and controlling the audio coder-decoder and the audio DSP;
the MCU calls a processing program in the memory, processes the input digital signal, identifies frequency components in the input digital signal, and processes the input digital signal to perform fast Fourier transform on the input digital signal to obtain frequency domain information of the input digital signal; that is, the frequency domain information of the digital signal is obtained by a first formula, where the first formula is:
F(ω)=FFT(f(N))
wherein, F (ω) is the frequency domain information of the input digital signal, F (n) is the input digital signal, and FFT is fast fourier transform;
the MCU reads the parameters of the electric tone basic tone controller, and according to the frequency components in the input digital signals, the expected basic tone and a preset adjusting method, calls a processing program in the memory to adjust the input digital signals to obtain output digital signals, namely the output digital signals are obtained through a second formula, wherein the second formula is as follows:
S(N)=IFFT(G(kω)·F(kω))
wherein, F (k ω) is the frequency domain information F (ω) of the input digital signal, k is the parameter value for the control of the electrical tone, G (k ω) is the amplitude adjustment function, s (n) is the output digital signal, IFFT is the inverse fast fourier transform.
For the amplitude adjustment function, the invention provides 2 preferred functions, G (k ω) 1 and
Figure BDA0001330359050000071
where H (ω) is the equal loudness curve for a certain sound pressure level, preferably 60 to 80 db. Selecting G (k ω) ═ 1 can reduce the difficulty of program implementation, and selection is performed
Figure BDA0001330359050000072
The sensitivity of human hearing to different frequencies can be taken into account and adjusted accordingly.
According to the method provided by the invention, the occupied storage space of the program can be smaller by carrying out parameterization processing on the program in the memory.
In one embodiment of the invention, processing an input digital signal comprises:
the MCU performs discrete Fourier transform on the input digital signal;
the MCU performs ideal filtering on the input digital signal after Fourier transform;
the MCU restores the filtered input digital signal to a time domain signal through inverse fourier transform.
According to the method provided by the invention, the ideal filtering processing of the frequency domain is carried out after Fourier transform, so that the processing of the audio signal can be more accurate.
In one embodiment of the invention, the MCU performs discrete fourier transform on the input digital signal, comprising:
the MCU performs windowing processing on the input digital signal, and the window function used for windowing is as follows:
Figure BDA0001330359050000081
where w (N) is the value of the window function, N is the frame length, and N is the sampling point.
According to the method provided by the invention, the specific window function is used, the input digital signal can be better processed, and the audio signal can be more accurately processed.
In an embodiment of the present invention, the MCU performs discrete fourier transform on the input digital signal, and may further be implemented as:
the MCU performs short-time Fourier transform or wavelet transform on the input digital signal.
According to the method provided by the invention, the problem of non-stationarity of the input digital signal can be solved, so that the audio signal is flexibly processed, and the audio signal is more accurately processed.
The invention also provides an audio signal processing system based on the electrical tone control, as shown in fig. 2, which is composed of an analog signal input terminal, an audio codec, an audio DSP, and an analog signal output terminal, and comprises:
the analog signal input end receives an input analog signal and sends the input analog signal to the audio codec from the analog signal input end;
the audio codec converts an input analog signal received from an analog signal input end into an input digital signal and sends the input digital signal to the audio DSP;
after receiving the input digital signal, the audio DSP identifies the frequency component in the input digital signal, adjusts the input digital signal according to the frequency component in the digital signal, an expected basic tone and a preset adjusting method to obtain an output digital signal, and sends the output digital signal to an audio codec;
the audio codec converts an output digital signal received from the audio DSP into an output analog signal and sends the output analog signal to an analog signal output end;
the analog signal output end outputs the received output analog signal.
In one embodiment of the present invention,
the audio DSP, including,
a memory storing a processing program for performing spectrum analysis on the digital signal and processing the digital signal according to the parameter;
an electric tone base controller comprising parameters for electric tone base control, said parameters comprising 13 values, representing A, A #, B, C, C #, D, D #, E, F, F #, G, G #, and #13 electric tone base;
the MCU comprises an initialization program and a control program and is used for initializing and controlling the audio coder-decoder and the audio DSP;
the MCU calls a processing program in the memory, processes the input digital signal, identifies frequency components in the input digital signal, and processes the input digital signal to perform fast Fourier transform on the input digital signal to obtain frequency domain information of the input digital signal; that is, the frequency domain information of the digital signal is obtained by a first formula, where the first formula is:
F(ω)=FFT(f(N))
wherein, F (ω) is the frequency domain information of the input digital signal, F (n) is the input digital signal, and FFT is fast fourier transform;
the MCU reads the parameters of the electric tone basic tone controller, and according to the frequency components in the input digital signals, the expected basic tone and a preset adjusting method, calls a processing program in the memory to adjust the input digital signals to obtain output digital signals, namely the output digital signals are obtained through a second formula, wherein the second formula is as follows:
S(N)=IFFT(G(kω)·F(kω))
wherein, F (k ω) is the frequency domain information F (ω) of the input digital signal, k is the parameter value for the control of the electrical tone, G (k ω) is the amplitude adjustment function, s (n) is the output digital signal, IFFT is the inverse fast fourier transform.
In one embodiment of the present invention, the MCU, processing the input digital signal, includes:
the MCU performs discrete Fourier transform on the input digital signal;
the MCU performs ideal filtering on the input digital signal after Fourier transform;
the MCU restores the filtered input digital signal to a time domain signal through inverse fourier transform.
In one embodiment of the present invention, the MCU performs discrete fourier transform on an input digital signal, including:
the MCU performs windowing processing on the input digital signal, and the window function used for windowing is as follows:
Figure BDA0001330359050000101
where w (N) is the value of the window function, N is the frame length, and N is the sampling point.
In an embodiment of the present invention, the MCU performs discrete fourier transform on the input digital signal, and may further implement:
the MCU performs short-time Fourier transform or wavelet transform on the input digital signal.
The audio signal processing method and system based on the electric tone pitch control can process the audio signal according to the electric tone pitch, occupy small storage space and process the audio signal more accurately.
As will be appreciated by one skilled in the art, embodiments of the present invention may be provided as a method, system, or computer program product. Accordingly, the present invention may take the form of an entirely hardware embodiment, an entirely software embodiment or an embodiment combining software and hardware aspects. Furthermore, the present invention may take the form of a computer program product embodied on one or more computer-usable storage media (including, but not limited to, disk storage, optical storage, and the like) having computer-usable program code embodied therein.
The present invention is described with reference to flowchart illustrations and/or block diagrams of methods, apparatus (systems), and computer program products according to embodiments of the invention. It will be understood that each flow and/or block of the flow diagrams and/or block diagrams, and combinations of flows and/or blocks in the flow diagrams and/or block diagrams, can be implemented by computer program instructions. These computer program instructions may be provided to a processor of a general purpose computer, special purpose computer, embedded processor, or other programmable data processing apparatus to produce a machine, such that the instructions, which execute via the processor of the computer or other programmable data processing apparatus, create means for implementing the functions specified in the flowchart flow or flows and/or block diagram block or blocks.
These computer program instructions may also be stored in a computer-readable memory that can direct a computer or other programmable data processing apparatus to function in a particular manner, such that the instructions stored in the computer-readable memory produce an article of manufacture including instruction means which implement the function specified in the flowchart flow or flows and/or block diagram block or blocks.
These computer program instructions may also be loaded onto a computer or other programmable data processing apparatus to cause a series of operational steps to be performed on the computer or other programmable apparatus to produce a computer implemented process such that the instructions which execute on the computer or other programmable apparatus provide steps for implementing the functions specified in the flowchart flow or flows and/or block diagram block or blocks.
It will be apparent to those skilled in the art that various changes and modifications may be made in the present invention without departing from the spirit and scope of the invention. Thus, if such modifications and variations of the present invention fall within the scope of the claims of the present invention and their equivalents, the present invention is also intended to include such modifications and variations.

Claims (8)

1. An audio signal processing method based on electric tone control is realized by an analog signal input end, an audio codec, an audio DSP and an analog signal output end, and is characterized by comprising the following steps:
the analog signal input end receives an input analog signal and sends the input analog signal to the audio codec from the analog signal input end;
the audio codec converts an input analog signal received from an analog signal input end into an input digital signal and sends the input digital signal to the audio DSP;
after receiving the input digital signal, the audio DSP identifies the frequency component in the input digital signal, adjusts the input digital signal according to the frequency component in the digital signal, an expected basic tone and a preset adjusting method to obtain an output digital signal, and sends the output digital signal to an audio codec;
the audio codec converts an output digital signal received from the audio DSP into an output analog signal and sends the output analog signal to an analog signal output end;
the analog signal output end outputs the received output analog signal;
the audio DSP comprises a Digital Signal Processor (DSP),
a memory storing a processing program for performing spectrum analysis on the digital signal and processing the digital signal according to the parameter;
an electric tone base controller comprising parameters for electric tone base control, said parameters comprising 13 values, representing A, A #, B, C, C #, D, D #, E, F, F #, G, G #, and #13 electric tone base;
the MCU comprises an initialization program and a control program and is used for initializing and controlling the audio coder-decoder and the audio DSP;
the MCU calls a processing program in the memory, processes the input digital signal, identifies frequency components in the input digital signal, and processes the input digital signal to perform fast Fourier transform on the input digital signal to obtain frequency domain information of the input digital signal; that is, the frequency domain information of the digital signal is obtained by a first formula, where the first formula is:
F(ω)=FFT(f(N))
wherein, F (ω) is the frequency domain information of the input digital signal, F (n) is the input digital signal, and FFT is fast fourier transform;
the MCU reads the parameters of the electric tone basic tone controller, and according to the frequency components in the input digital signals, the expected basic tone and a preset adjusting method, calls a processing program in the memory to adjust the input digital signals to obtain output digital signals, namely the output digital signals are obtained through a second formula, wherein the second formula is as follows:
S(N)=IFFT(G(kω)·F(kω))
wherein, F (k ω) is the frequency domain information F (ω) of the input digital signal, k is the parameter value for the control of the electrical tone, G (k ω) is the amplitude adjustment function, s (n) is the output digital signal, IFFT is the inverse fast fourier transform.
2. The method of claim 1, wherein the processing the input digital signal comprises:
the MCU performs discrete Fourier transform on the input digital signal;
the MCU performs ideal filtering on the input digital signal after Fourier transform;
the MCU restores the filtered input digital signal to a time domain signal through inverse fourier transform.
3. The method of claim 2, wherein the MCU performs a discrete fourier transform on an input digital signal, comprising:
the MCU performs windowing processing on the input digital signal, and the window function used for windowing is as follows:
Figure FDA0002409617010000031
where w (N) is the value of the window function, N is the frame length, and N is the sampling point.
4. The method of claim 2, wherein the MCU performs a discrete fourier transform on the input digital signal, further operable to:
the MCU performs short-time Fourier transform or wavelet transform on the input digital signal.
5. An audio signal processing system based on electric tone control, which comprises an analog signal input end, an audio codec, an audio DSP and an analog signal output end, and is characterized by comprising:
the analog signal input end receives an input analog signal and sends the input analog signal to the audio codec from the analog signal input end;
the audio codec converts an input analog signal received from an analog signal input end into an input digital signal and sends the input digital signal to the audio DSP;
after receiving the input digital signal, the audio DSP identifies the frequency component in the input digital signal, adjusts the input digital signal according to the frequency component in the digital signal, an expected basic tone and a preset adjusting method to obtain an output digital signal, and sends the output digital signal to an audio codec;
the audio codec converts an output digital signal received from the audio DSP into an output analog signal and sends the output analog signal to an analog signal output end;
the analog signal output end outputs the received output analog signal;
the audio DSP, including,
a memory storing a processing program for performing spectrum analysis on the digital signal and processing the digital signal according to the parameter;
an electric tone base controller comprising parameters for electric tone base control, said parameters comprising 13 values, representing A, A #, B, C, C #, D, D #, E, F, F #, G, G #, and #13 electric tone base;
the MCU comprises an initialization program and a control program and is used for initializing and controlling the audio coder-decoder and the audio DSP;
the MCU calls a processing program in the memory, processes the input digital signal, identifies frequency components in the input digital signal, and processes the input digital signal to perform fast Fourier transform on the input digital signal to obtain frequency domain information of the input digital signal; that is, the frequency domain information of the digital signal is obtained by a first formula, where the first formula is:
F(ω)=FFT(f(N))
wherein, F (ω) is the frequency domain information of the input digital signal, F (n) is the input digital signal, and FFT is fast fourier transform;
the MCU reads the parameters of the electric tone basic tone controller, and according to the frequency components in the input digital signals, the expected basic tone and a preset adjusting method, calls a processing program in the memory to adjust the input digital signals to obtain output digital signals, namely the output digital signals are obtained through a second formula, wherein the second formula is as follows:
S(N)=IFFT(G(kω)·F(kω))
wherein, F (k ω) is the frequency domain information F (ω) of the input digital signal, k is the parameter value for the control of the electrical tone, G (k ω) is the amplitude adjustment function, s (n) is the output digital signal, IFFT is the inverse fast fourier transform.
6. The system of claim 5, wherein the MCU processes the input digital signal, comprising:
the MCU performs discrete Fourier transform on the input digital signal;
the MCU performs ideal filtering on the input digital signal after Fourier transform;
the MCU restores the filtered input digital signal to a time domain signal through inverse fourier transform.
7. The system of claim 6, wherein the MCU, performing a discrete fourier transform on the input digital signal, comprises:
the MCU performs windowing processing on the input digital signal, and the window function used for windowing is as follows:
Figure FDA0002409617010000051
where w (N) is the value of the window function, N is the frame length, and N is the sampling point.
8. The system of claim 6, wherein the MCU, performing a discrete fourier transform on the input digital signal, is further operable to:
the MCU performs short-time Fourier transform or wavelet transform on the input digital signal.
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