CN107481727A - A kind of acoustic signal processing method and system based on the control of electric sound keynote - Google Patents
A kind of acoustic signal processing method and system based on the control of electric sound keynote Download PDFInfo
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- G—PHYSICS
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- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Speech 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
- G10L21/003—Changing voice quality, e.g. pitch or formants
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- G10L21/013—Adapting to target pitch
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech 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
- G10L19/04—Speech 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 a kind of acoustic signal processing method based on the control of electric sound keynote and system, the processing method to be：Input end of analog signal receives input analog signal, and input analog signal is sent into audio codec from input end of analog signal；The input analog signal received from input end of analog signal is converted into input data signal by audio codec, and input data signal is sent into audio DSP；After audio DSP receives input data signal, frequency content in identified input data signal, frequency content, desired keynote and default method of adjustment in data signal are adjusted to input data signal, output digit signals are obtained, and output digit signals are sent to audio codec；The output digit signals received from audio DSP are converted into output analog signal by audio codec, and output analog signal is sent into analog signal output；Analog signal output is exported according to the output analog signal received.
Description
Technical field
The present invention relates to acoustic processing field, more particularly to a kind of acoustic signal processing method based on the control of electric sound keynote
And system.
Background technology
The prevalence of electronic audio signal at present so that the processing to electronic audio signal has extensive demand, and at it
In, the electric sound keynote of audio signal as desired is carried out into sharp, flat adjustment etc. turns into more urgent in Audio Signal Processing
A kind of demand.Although thering are some equipment, software to handle audio signal now, audio letter can only be substantially calculated
Number keynote and the keynote of audio signal desirably can not be adjusted to audio signal.Therefore need one kind being capable of basis
The method and system that electric sound keynote is handled audio signal.
The content of the invention
To solve problem above, the present invention provides a kind of acoustic signal processing method based on the control of electric sound keynote and is
System.
A kind of acoustic signal processing method based on the control of electric sound keynote provided by the invention, by input end of analog signal,
Audio codec, audio DSP (digital signal processor, digital signal processor), analog signal output
Realize, including：
Input end of analog signal receives input analog signal, and input analog signal is sent to from input end of analog signal
Audio codec；
The input analog signal received from input end of analog signal is converted into input data signal by audio codec,
And input data signal is sent to audio DSP；
After audio DSP receives input data signal, the frequency content in identified input data signal, according to data signal
In frequency content, desired keynote and default method of adjustment to input data signal be adjusted, obtain exporting digital letter
Number, and output digit signals are sent to audio codec；
The output digit signals received from audio DSP are converted into output analog signal by audio codec, and will be defeated
Go out analog signal and be sent to analog signal output；
Analog signal output is exported according to the output analog signal received.
Preferably,
The audio DSP, including,
Memory, is stored with processing routine, and the processing routine is used to carry out data signal spectrum analysis and according to ginseng
It is several that data signal is handled；
Electric sound keynote controller, comprising the parameter controlled for electric sound keynote, the parameter includes 13 and is worth, expression A,
A#, B, C, C#, D, D#, E, F, F#, G, G#, #13 electric sound keynotes；
MCU (MicroControllerUnit, micro-control unit, i.e. single-chip microcomputer), contain initialization program and control journey
Sequence, for carrying out initialization and control to audio codec and audio DSP；
MCU calls the processing routine in memory, and input data signal is handled, and identifies in input data signal
Frequency content, it is to carry out Fast Fourier Transform (FFT) to the data signal of input that described pair of input data signal, which carries out processing, is obtained
To the frequency domain information of input data signal；The frequency domain information of data signal is obtained by the first formula, first formula is：
F (ω)=FFT (f (N))
Wherein, F (ω) is the frequency domain information of input data signal, and f (N) is input data signal, and FFT is fast Fourier
Conversion；
MCU reads the parameter of electric sound keynote controller, frequency content, desired base in the input data signal
Reconcile default method of adjustment, call the processing routine in memory to be adjusted input data signal, obtain output numeral
Signal, i.e., obtain output digit signals by the second formula, and second formula is：
S (N)=IFFT (G (k ω) F (k ω))
Wherein, F (k ω) is stretchings of the frequency domain information F (ω) in frequency of input data signal, and k is for electric sound
The parameter value of keynote control, G (k ω) is range-adjusting function, and S (N) is output digit signals, and IFFT is fast Fourier transforma
Change.
Preferably, described pair of input data signal is handled, including：
MCU carries out discrete Fourier transform to input data signal；
MCU carries out ideal filtering to the input data signal after Fourier transformation；
Filtered input data signal is reduced to time-domain signal by MCU by inversefouriertransform.
Preferably, the MCU carries out discrete Fourier transform to input data signal, including：
MCU carries out windowing process to input data signal, and the window function used in adding window is：
Wherein, w (n) is the value of window function, and N is frame length, and n is sampled point.
Preferably, the MCU carries out discrete Fourier transform to input data signal, can also be embodied as：
MCU carries out Short Time Fourier Transform or wavelet transformation to input data signal.
Present invention also offers a kind of audio signal processing based on the control of electric sound keynote, inputted by analog signal
End, audio codec, audio DSP, analog signal output composition, including：
Input end of analog signal receives input analog signal, and input analog signal is sent to from input end of analog signal
Audio codec；
The input analog signal received from input end of analog signal is converted into input data signal by audio codec,
And input data signal is sent to audio DSP；
After audio DSP receives input data signal, the frequency content in identified input data signal, according to data signal
In frequency content, desired keynote and default method of adjustment to input data signal be adjusted, obtain exporting digital letter
Number, and output digit signals are sent to audio codec；
The output digit signals received from audio DSP are converted into output analog signal by audio codec, and will be defeated
Go out analog signal and be sent to analog signal output；
Analog signal output is exported according to the output analog signal received.
Preferably,
Audio DSP, including,
Memory, is stored with processing routine, and the processing routine is used to carry out data signal spectrum analysis and according to ginseng
It is several that data signal is handled；
Electric sound keynote controller, comprising the parameter controlled for electric sound keynote, the parameter includes 13 and is worth, expression A,
A#, B, C, C#, D, D#, E, F, F#, G, G#, #13 electric sound keynotes；
MCU, containing initialization program and control program, for audio codec and audio DSP are carried out initialization and
Control；
MCU calls the processing routine in memory, and input data signal is handled, and identifies in input data signal
Frequency content, it is to carry out Fast Fourier Transform (FFT) to the data signal of input that described pair of input data signal, which carries out processing, is obtained
To the frequency domain information of input data signal；The frequency domain information of data signal is obtained by the first formula, first formula is：
F (ω)=FFT (f (N))
Wherein, F (ω) is the frequency domain information of input data signal, and f (N) is input data signal, and FFT is fast Fourier
Conversion；
MCU reads the parameter of electric sound keynote controller, frequency content, desired base in the input data signal
Reconcile default method of adjustment, call the processing routine in memory to be adjusted input data signal, obtain output numeral
Signal, i.e., obtain output digit signals by the second formula, and second formula is：
S (N)=IFFT (G (k ω) F (k ω))
Wherein, F (k ω) is stretchings of the frequency domain information F (ω) in frequency of input data signal, and k is for electric sound
The parameter value of keynote control, G (k ω) is range-adjusting function, and S (N) is output digit signals, and IFFT is fast Fourier transforma
Change.
Preferably, MCU, input data signal is handled, including：
MCU carries out discrete Fourier transform to input data signal；
MCU carries out ideal filtering to the input data signal after Fourier transformation；
Filtered input data signal is reduced to time-domain signal by MCU by inversefouriertransform.
Preferably, MCU, discrete Fourier transform is carried out to input data signal, including：
MCU carries out windowing process to input data signal, and the window function used in adding window is：
Wherein, w (n) is the value of window function, and N is frame length, and n is sampled point.
Preferably, MCU, discrete Fourier transform is carried out to input data signal, can be also embodied as：
MCU carries out Short Time Fourier Transform or wavelet transformation to input data signal.
Some beneficial effects of the present invention can include：
A kind of acoustic signal processing method and system based on the control of electric sound keynote provided by the invention, can be according to electric sound
Keynote is handled audio signal, and occupancy memory space is small, and the processing to audio signal is more accurate.
Other features and advantages of the present invention will be illustrated in the following description, also, partly becomes from specification
Obtain it is clear that or being understood by implementing the present invention.The purpose of the present invention and other advantages can be by the explanations write
Specifically noted structure is realized and obtained in book, claims and accompanying drawing.
Below by drawings and examples, technical scheme is described in further detail.
Brief description of the drawings
Accompanying drawing is used for providing a further understanding of the present invention, and a part for constitution instruction, the reality with the present invention
Apply example to be used to explain the present invention together, be not construed as limiting the invention.In the accompanying drawings：
Fig. 1 is a kind of flow chart of the acoustic signal processing method based on the control of electric sound keynote in the embodiment of the present invention；
Fig. 2 is a kind of schematic diagram of the audio signal processing based on the control of electric sound keynote in the embodiment of the present invention.
Embodiment
The preferred embodiments of the present invention are illustrated below in conjunction with accompanying drawing, it will be appreciated that described herein preferred real
Apply example to be merely to illustrate and explain the present invention, be not intended to limit the present invention.
Fig. 1 is a kind of flow chart of the acoustic signal processing method based on the control of electric sound keynote in the embodiment of the present invention.Such as
Shown in Fig. 1, this method is realized by input end of analog signal, audio codec, audio DSP, analog signal output, including：
Step S101, input end of analog signal receives input analog signal, and it is defeated from analog signal to input analog signal
Enter end and be sent to audio codec；
Step S102, the input analog signal received from input end of analog signal is converted into input by audio codec
Data signal, and input data signal is sent to audio DSP；
Step S103, after audio DSP receives input data signal, the frequency content in identified input data signal, root
Input data signal is adjusted according to the frequency content in data signal, desired keynote and default method of adjustment, obtained
Output digit signals, and output digit signals are sent to audio codec；
Step S104, the output digit signals received from audio DSP are converted into output simulation letter by audio codec
Number, and output analog signal is sent to analog signal output；
Step S105, analog signal output is exported according to the output analog signal received.
According to method provided by the invention, audio signal can be handled according to electric sound keynote.
In one embodiment of the invention,
Audio DSP, including,
Memory, is stored with processing routine, and the processing routine is used to carry out data signal spectrum analysis and according to ginseng
It is several that data signal is handled；
Electric sound keynote controller, comprising the parameter controlled for electric sound keynote, the parameter includes 13 and is worth, expression A,
A#, B, C, C#, D, D#, E, F, F#, G, G#, #13 electric sound keynotes；
MCU, containing initialization program and control program, for audio codec and audio DSP are carried out initialization and
Control；
MCU calls the processing routine in memory, and input data signal is handled, and identifies in input data signal
Frequency content, it is to carry out Fast Fourier Transform (FFT) to the data signal of input that described pair of input data signal, which carries out processing, is obtained
To the frequency domain information of input data signal；The frequency domain information of data signal is obtained by the first formula, first formula is：
F (ω)=FFT (f (N))
Wherein, F (ω) is the frequency domain information of input data signal, and f (N) is input data signal, and FFT is fast Fourier
Conversion；
MCU reads the parameter of electric sound keynote controller, frequency content, desired base in the input data signal
Reconcile default method of adjustment, call the processing routine in memory to be adjusted input data signal, obtain output numeral
Signal, i.e., obtain output digit signals by the second formula, and second formula is：
S (N)=IFFT (G (k ω) F (k ω))
Wherein, F (k ω) is stretchings of the frequency domain information F (ω) in frequency of input data signal, and k is for electric sound
The parameter value of keynote control, G (k ω) is range-adjusting function, and S (N) is output digit signals, and IFFT is fast Fourier transforma
Change.
To range-adjusting function, the present invention provides 2 preferred functions, G (k ω)=1 HeWherein H
(ω) is equal loudness contour corresponding to a certain sound pressure level, equal loudness contour preferably corresponding to 60 to 80 decibels of sound pressure level.Select G (k
It can ω)=1 reduce the difficulty of program realization, selectionIt can be considered that mankind's hearing is to the quick of different frequency
Sensitivity is so as to making corresponding adjustment.
According to method provided by the invention, by the parameterized treatment to memory Program, the occupancy of program can be made
Memory space is smaller.
In one embodiment of the invention, input data signal is handled, including：
MCU carries out discrete Fourier transform to input data signal；
MCU carries out ideal filtering to the input data signal after Fourier transformation；
Filtered input data signal is reduced to time-domain signal by MCU by inversefouriertransform.
According to method provided by the invention, handled by the ideal filtering in the laggard line frequency domain of Fourier transformation, can be right
The processing of audio signal is more accurate.
In one embodiment of the invention, MCU carries out discrete Fourier transform to input data signal, including：
MCU carries out windowing process to input data signal, and the window function used in adding window is：
Wherein, w (n) is the value of window function, and N is frame length, and n is sampled point.
, can be preferably to inputting at data signal using specific window function according to method provided by the invention
Reason, so as to which the processing to audio signal is more accurate.
In one embodiment of the invention, the MCU carries out discrete Fourier transform to input data signal, can also be real
Shi Wei：
MCU carries out Short Time Fourier Transform or wavelet transformation to input data signal.
According to method provided by the invention, can solve the problem that input data signal it is non-stationary the problem of, so as to flexible
Audio signal is handled, and then the processing to audio signal is more accurate.
Present invention also offers a kind of audio signal processing based on the control of electric sound keynote, as shown in Fig. 2 by simulating
Signal input part, audio codec, audio DSP, analog signal output composition, including：
Input end of analog signal receives input analog signal, and input analog signal is sent to from input end of analog signal
Audio codec；
The input analog signal received from input end of analog signal is converted into input data signal by audio codec,
And input data signal is sent to audio DSP；
After audio DSP receives input data signal, the frequency content in identified input data signal, according to data signal
In frequency content, desired keynote and default method of adjustment to input data signal be adjusted, obtain exporting digital letter
Number, and output digit signals are sent to audio codec；
The output digit signals received from audio DSP are converted into output analog signal by audio codec, and will be defeated
Go out analog signal and be sent to analog signal output；
Analog signal output is exported according to the output analog signal received.
In one embodiment of the invention,
Audio DSP, including,
Memory, is stored with processing routine, and the processing routine is used to carry out data signal spectrum analysis and according to ginseng
It is several that data signal is handled；
Electric sound keynote controller, comprising the parameter controlled for electric sound keynote, the parameter includes 13 and is worth, expression A,
A#, B, C, C#, D, D#, E, F, F#, G, G#, #13 electric sound keynotes；
MCU, containing initialization program and control program, for audio codec and audio DSP are carried out initialization and
Control；
MCU calls the processing routine in memory, and input data signal is handled, and identifies in input data signal
Frequency content, it is to carry out Fast Fourier Transform (FFT) to the data signal of input that described pair of input data signal, which carries out processing, is obtained
To the frequency domain information of input data signal；The frequency domain information of data signal is obtained by the first formula, first formula is：
F (ω)=FFT (f (N))
Wherein, F (ω) is the frequency domain information of input data signal, and f (N) is input data signal, and FFT is fast Fourier
Conversion；
MCU reads the parameter of electric sound keynote controller, frequency content, desired base in the input data signal
Reconcile default method of adjustment, call the processing routine in memory to be adjusted input data signal, obtain output numeral
Signal, i.e., obtain output digit signals by the second formula, and second formula is：
S (N)=IFFT (G (k ω) F (k ω))
Wherein, F (k ω) is stretchings of the frequency domain information F (ω) in frequency of input data signal, and k is for electric sound
The parameter value of keynote control, G (k ω) is range-adjusting function, and S (N) is output digit signals, and IFFT is fast Fourier transforma
Change.
In one embodiment of the invention, MCU, input data signal is handled, including：
MCU carries out discrete Fourier transform to input data signal；
MCU carries out ideal filtering to the input data signal after Fourier transformation；
Filtered input data signal is reduced to time-domain signal by MCU by inversefouriertransform.
In one embodiment of the invention, MCU, discrete Fourier transform is carried out to input data signal, including：
MCU carries out windowing process to input data signal, and the window function used in adding window is：
Wherein, w (n) is the value of window function, and N is frame length, and n is sampled point.
In one embodiment of the invention, MCU, discrete Fourier transform is carried out to input data signal, can also be implemented
For：
MCU carries out Short Time Fourier Transform or wavelet transformation to input data signal.
A kind of acoustic signal processing method and system based on the control of electric sound keynote provided by the invention, can be according to electric sound
Keynote is handled audio signal, and occupancy memory space is small, and the processing to audio signal is more accurate.
It should be understood by those skilled in the art that, embodiments of the invention can be provided as method, system or computer program
Product.Therefore, the present invention can use the reality in terms of complete hardware embodiment, complete software embodiment or combination software and hardware
Apply the form of example.Moreover, the present invention can use the computer for wherein including computer usable program code in one or more
The shape for the computer program product that usable storage medium is implemented on (including but is not limited to magnetic disk storage and optical memory etc.)
Formula.
The present invention is the flow with reference to method according to embodiments of the present invention, equipment (system) and computer program product
Figure and/or block diagram describe.It should be understood that can be by every first-class in computer program instructions implementation process figure and/or block diagram
Journey and/or the flow in square frame and flow chart and/or block diagram and/or the combination of square frame.These computer programs can be provided
The processors of all-purpose computer, special-purpose computer, Embedded Processor or other programmable data processing devices is instructed to produce
A raw machine so that produced by the instruction of computer or the computing device of other programmable data processing devices for real
The device for the function of being specified in present one flow of flow chart or one square frame of multiple flows and/or block diagram or multiple square frames.
These computer program instructions, which may be alternatively stored in, can guide computer or other programmable data processing devices with spy
Determine in the computer-readable memory that mode works so that the instruction being stored in the computer-readable memory, which produces, to be included referring to
Make the manufacture of device, the command device realize in one flow of flow chart or multiple flows and/or one square frame of block diagram or
The function of being specified in multiple square frames.
These computer program instructions can be also loaded into computer or other programmable data processing devices so that counted
Series of operation steps is performed on calculation machine or other programmable devices to produce computer implemented processing, so as in computer or
The instruction performed on other programmable devices is provided for realizing in one flow of flow chart or multiple flows and/or block diagram one
The step of function of being specified in individual square frame or multiple square frames.
Obviously, those skilled in the art can carry out the essence of various changes and modification without departing from the present invention to the present invention
God and scope.So, if these modifications and variations of the present invention belong to the scope of the claims in the present invention and its equivalent technologies
Within, then the present invention is also intended to comprising including these changes and modification.
Claims (10)
- A kind of 1. acoustic signal processing method based on the control of electric sound keynote, by input end of analog signal, audio codec, sound Frequency DSP, analog signal output are realized, it is characterised in that including：Input end of analog signal receives input analog signal, and input analog signal is sent into audio from input end of analog signal Codec；The input analog signal received from input end of analog signal is converted into input data signal by audio codec, and will Input data signal is sent to audio DSP；After audio DSP receives input data signal, the frequency content in identified input data signal, according in data signal Frequency content, desired keynote and default method of adjustment are adjusted to input data signal, obtain output digit signals, and Output digit signals are sent to audio codec；The output digit signals received from audio DSP are converted into output analog signal by audio codec, and will export mould Intend signal and be sent to analog signal output；Analog signal output is exported according to the output analog signal received.
- 2. the method as described in claim 1, it is characterised in that：The audio DSP, including,Memory, is stored with processing routine, and the processing routine is used to carry out data signal spectrum analysis and according to parameter pair Data signal is handled；Electric sound keynote controller, comprising the parameter controlled for electric sound keynote, the parameter includes 13 values, represent A, A#, B, C, C#, D, D#, E, F, F#, G, G#, #13 electric sound keynotes；MCU, containing initialization program and control program, for carrying out initialization and control to audio codec and audio DSP；MCU calls the processing routine in memory, and input data signal is handled, and identifies the frequency in input data signal Rate composition, it is to carry out Fast Fourier Transform (FFT) to the data signal of input that described pair of input data signal, which carries out processing, is obtained defeated Enter the frequency domain information of data signal；The frequency domain information of data signal is obtained by the first formula, first formula is：F (ω)=FFT (f (N))Wherein, F (ω) is the frequency domain information of input data signal, and f (N) is input data signal, and FFT becomes for fast Fourier Change；MCU reads the parameter of electric sound keynote controller, frequency content, desired keynote in the input data signal and Default method of adjustment, call the processing routine in memory to be adjusted input data signal, obtain output digit signals, Output digit signals are obtained by the second formula, second formula is：S (N)=IFFT (G (k ω) F (k ω))Wherein, F (k ω) is stretchings of the frequency domain information F (ω) in frequency of input data signal, and k is for electric sound keynote The parameter value of control, G (k ω) are range-adjusting function, and S (N) is output digit signals, and IFFT is inverse fast Fourier transform.
- 3. method as claimed in claim 2, it is characterised in that described pair of input data signal is handled, including：MCU carries out discrete Fourier transform to input data signal；MCU carries out ideal filtering to the input data signal after Fourier transformation；Filtered input data signal is reduced to time-domain signal by MCU by inversefouriertransform.
- 4. method as claimed in claim 3, it is characterised in that the MCU carries out discrete fourier change to input data signal Change, including：MCU carries out windowing process to input data signal, and the window function used in adding window is：<mrow> <mi>w</mi> <mrow> <mo>(</mo> <mi>n</mi> <mo>)</mo> </mrow> <mo>=</mo> <mfenced open = "{" close = ""> <mtable> <mtr> <mtd> <mrow> <mn>0.41</mn> <mo>-</mo> <mn>0.37</mn> <mi>c</mi> <mi>o</mi> <mi>s</mi> <mo>&lsqb;</mo> <mfrac> <mrow> <mn>2</mn> <mi>&pi;</mi> <mrow> <mo>(</mo> <mi>n</mi> <mo>-</mo> <mn>1</mn> <mo>)</mo> </mrow> </mrow> <mrow> <mi>N</mi> <mo>-</mo> <mn>1</mn> </mrow> </mfrac> <mo>&rsqb;</mo> <mo>+</mo> <mn>0.22</mn> <mi>s</mi> <mi>i</mi> <mi>n</mi> <mo>&lsqb;</mo> <mfrac> <mrow> <mi>&pi;</mi> <mrow> <mo>(</mo> <mi>n</mi> <mo>-</mo> <mn>1</mn> <mo>)</mo> </mrow> </mrow> <mrow> <mi>N</mi> <mo>-</mo> <mn>1</mn> </mrow> </mfrac> <mo>&rsqb;</mo> <mo>,</mo> </mrow> </mtd> <mtd> <mrow> <mn>1</mn> <mo>&le;</mo> <mi>n</mi> <mo>&le;</mo> <mfrac> <mi>N</mi> <mn>2</mn> </mfrac> </mrow> </mtd> </mtr> <mtr> <mtd> <mrow> <mn>0.55</mn> <mo>-</mo> <mn>0.45</mn> <mi>c</mi> <mi>o</mi> <mi>s</mi> <mo>&lsqb;</mo> <mfrac> <mrow> <mn>2</mn> <mi>&pi;</mi> <mrow> <mo>(</mo> <mi>n</mi> <mo>-</mo> <mn>1</mn> <mo>)</mo> </mrow> </mrow> <mrow> <mi>N</mi> <mo>-</mo> <mn>1</mn> </mrow> </mfrac> <mo>&rsqb;</mo> <mo>,</mo> </mrow> </mtd> <mtd> <mrow> <mfrac> <mrow> <mi>N</mi> <mo>+</mo> <mn>1</mn> </mrow> <mn>2</mn> </mfrac> <mo>&le;</mo> <mi>n</mi> <mo>&le;</mo> <mi>N</mi> </mrow> </mtd> </mtr> <mtr> <mtd> <mn>0</mn> </mtd> <mtd> <mrow> <mi>n</mi> <mo>=</mo> <mi>e</mi> <mi>l</mi> <mi>s</mi> <mi>e</mi> </mrow> </mtd> </mtr> </mtable> </mfenced> </mrow>Wherein, w (n) is the value of window function, and N is frame length, and n is sampled point.
- 5. method as claimed in claim 3, it is characterised in that the MCU carries out discrete fourier change to input data signal Change, can also be embodied as：MCU carries out Short Time Fourier Transform or wavelet transformation to input data signal.
- A kind of 6. audio signal processing based on the control of electric sound keynote, by input end of analog signal, audio codec, sound Frequency DSP, analog signal output composition, it is characterised in that including：Input end of analog signal receives input analog signal, and input analog signal is sent into audio from input end of analog signal Codec；The input analog signal received from input end of analog signal is converted into input data signal by audio codec, and will Input data signal is sent to audio DSP；After audio DSP receives input data signal, the frequency content in identified input data signal, according in data signal Frequency content, desired keynote and default method of adjustment are adjusted to input data signal, obtain output digit signals, and Output digit signals are sent to audio codec；The output digit signals received from audio DSP are converted into output analog signal by audio codec, and will export mould Intend signal and be sent to analog signal output；Analog signal output is exported according to the output analog signal received.
- 7. system as claimed in claim 6, it is characterised in that：Audio DSP, including,Memory, is stored with processing routine, and the processing routine is used to carry out data signal spectrum analysis and according to parameter pair Data signal is handled；Electric sound keynote controller, comprising the parameter controlled for electric sound keynote, the parameter includes 13 values, represent A, A#, B, C, C#, D, D#, E, F, F#, G, G#, #13 electric sound keynotes；MCU, containing initialization program and control program, for carrying out initialization and control to audio codec and audio DSP；MCU calls the processing routine in memory, and input data signal is handled, and identifies the frequency in input data signal Rate composition, it is to carry out Fast Fourier Transform (FFT) to the data signal of input that described pair of input data signal, which carries out processing, is obtained defeated Enter the frequency domain information of data signal；The frequency domain information of data signal is obtained by the first formula, first formula is：F (ω)=FFT (f (N))Wherein, F (ω) is the frequency domain information of input data signal, and f (N) is input data signal, and FFT becomes for fast Fourier Change；MCU reads the parameter of electric sound keynote controller, frequency content, desired keynote in the input data signal and Default method of adjustment, call the processing routine in memory to be adjusted input data signal, obtain output digit signals, Output digit signals are obtained by the second formula, second formula is：S (N)=IFFT (G (k ω) F (k ω))Wherein, F (k ω) is stretchings of the frequency domain information F (ω) in frequency of input data signal, and k is for electric sound keynote The parameter value of control, G (k ω) are range-adjusting function, and S (N) is output digit signals, and IFFT is inverse fast Fourier transform.
- 8. system as claimed in claim 7, it is characterised in that MCU, input data signal is handled, including：MCU carries out discrete Fourier transform to input data signal；MCU carries out ideal filtering to the input data signal after Fourier transformation；Filtered input data signal is reduced to time-domain signal by MCU by inversefouriertransform.
- 9. system as claimed in claim 8, it is characterised in that MCU, discrete Fourier transform is carried out to input data signal, Including：MCU carries out windowing process to input data signal, and the window function used in adding window is：<mrow> <mi>w</mi> <mrow> <mo>(</mo> <mi>n</mi> <mo>)</mo> </mrow> <mo>=</mo> <mfenced open = "{" close = ""> <mtable> <mtr> <mtd> <mrow> <mn>0.41</mn> <mo>-</mo> <mn>0.37</mn> <mi>c</mi> <mi>o</mi> <mi>s</mi> <mo>&lsqb;</mo> <mfrac> <mrow> <mn>2</mn> <mi>&pi;</mi> <mrow> <mo>(</mo> <mi>n</mi> <mo>-</mo> <mn>1</mn> <mo>)</mo> </mrow> </mrow> <mrow> <mi>N</mi> <mo>-</mo> <mn>1</mn> </mrow> </mfrac> <mo>&rsqb;</mo> <mo>+</mo> <mn>0.22</mn> <mi>s</mi> <mi>i</mi> <mi>n</mi> <mo>&lsqb;</mo> <mfrac> <mrow> <mi>&pi;</mi> <mrow> <mo>(</mo> <mi>n</mi> <mo>-</mo> <mn>1</mn> <mo>)</mo> </mrow> </mrow> <mrow> <mi>N</mi> <mo>-</mo> <mn>1</mn> </mrow> </mfrac> <mo>&rsqb;</mo> <mo>,</mo> </mrow> </mtd> <mtd> <mrow> <mn>1</mn> <mo>&le;</mo> <mi>n</mi> <mo>&le;</mo> <mfrac> <mi>N</mi> <mn>2</mn> </mfrac> </mrow> </mtd> </mtr> <mtr> <mtd> <mrow> <mn>0.55</mn> <mo>-</mo> <mn>0.45</mn> <mi>c</mi> <mi>o</mi> <mi>s</mi> <mo>&lsqb;</mo> <mfrac> <mrow> <mn>2</mn> <mi>&pi;</mi> <mrow> <mo>(</mo> <mi>n</mi> <mo>-</mo> <mn>1</mn> <mo>)</mo> </mrow> </mrow> <mrow> <mi>N</mi> <mo>-</mo> <mn>1</mn> </mrow> </mfrac> <mo>&rsqb;</mo> <mo>,</mo> </mrow> </mtd> <mtd> <mrow> <mfrac> <mrow> <mi>N</mi> <mo>+</mo> <mn>1</mn> </mrow> <mn>2</mn> </mfrac> <mo>&le;</mo> <mi>n</mi> <mo>&le;</mo> <mi>N</mi> </mrow> </mtd> </mtr> <mtr> <mtd> <mn>0</mn> </mtd> <mtd> <mrow> <mi>n</mi> <mo>=</mo> <mi>e</mi> <mi>l</mi> <mi>s</mi> <mi>e</mi> </mrow> </mtd> </mtr> </mtable> </mfenced> </mrow>Wherein, w (n) is the value of window function, and N is frame length, and n is sampled point.
- 10. system as claimed in claim 8, it is characterised in that MCU, discrete Fourier transform is carried out to input data signal, It can also be embodied as：MCU carries out Short Time Fourier Transform or wavelet transformation to input data signal.
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Cited By (5)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN111698133A (en) * | 2020-06-23 | 2020-09-22 | 上海仁童电子科技有限公司 | Waveform correction instrument and waveform correction method |
| CN111767020A (en) * | 2020-05-18 | 2020-10-13 | 深圳市东微智能科技股份有限公司 | Method, device and terminal for optimizing audio processing and readable storage medium |
| CN112201263A (en) * | 2020-10-16 | 2021-01-08 | 广州资云科技有限公司 | Electric tone adjusting system based on song recognition |
| CN113470670A (en) * | 2021-06-30 | 2021-10-01 | 广州资云科技有限公司 | Method and system for quickly switching tone of electric tone |
| CN114449339A (en) * | 2022-02-16 | 2022-05-06 | 深圳万兴软件有限公司 | Background sound effect conversion method and device, computer equipment and storage medium |
Citations (6)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN1178964A (en) * | 1996-07-08 | 1998-04-15 | 索尼公司 | Sound signal processor |
| CN101179278A (en) * | 2006-11-07 | 2008-05-14 | 扬智科技股份有限公司 | Acoustics system and voice signal coding method thereof |
| JP2009071776A (en) * | 2007-09-18 | 2009-04-02 | Sanyo Electric Co Ltd | Tone adjusting circuit |
| US20090093896A1 (en) * | 2007-10-05 | 2009-04-09 | Yoshiyuki Kobayashi | Signal Processing Apparatus, Signal Processing Method, and Program Therefor |
| CN101425004A (en) * | 2007-10-29 | 2009-05-06 | 福建星网锐捷通讯股份有限公司 | Method and device for using voice as input control of computer system |
| CN201243266Y (en) * | 2008-05-16 | 2009-05-20 | 华南理工大学 | Equilibrium processing device for frequency response characteristic of audio-frequency system based on DSP |
-
2017
- 2017-06-23 CN CN201710485440.4A patent/CN107481727B/en active Active
Patent Citations (6)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN1178964A (en) * | 1996-07-08 | 1998-04-15 | 索尼公司 | Sound signal processor |
| CN101179278A (en) * | 2006-11-07 | 2008-05-14 | 扬智科技股份有限公司 | Acoustics system and voice signal coding method thereof |
| JP2009071776A (en) * | 2007-09-18 | 2009-04-02 | Sanyo Electric Co Ltd | Tone adjusting circuit |
| US20090093896A1 (en) * | 2007-10-05 | 2009-04-09 | Yoshiyuki Kobayashi | Signal Processing Apparatus, Signal Processing Method, and Program Therefor |
| CN101425004A (en) * | 2007-10-29 | 2009-05-06 | 福建星网锐捷通讯股份有限公司 | Method and device for using voice as input control of computer system |
| CN201243266Y (en) * | 2008-05-16 | 2009-05-20 | 华南理工大学 | Equilibrium processing device for frequency response characteristic of audio-frequency system based on DSP |
Cited By (9)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN111767020A (en) * | 2020-05-18 | 2020-10-13 | 深圳市东微智能科技股份有限公司 | Method, device and terminal for optimizing audio processing and readable storage medium |
| CN111767020B (en) * | 2020-05-18 | 2024-05-17 | 深圳市东微智能科技股份有限公司 | Optimized audio processing method, device, terminal and readable storage medium |
| CN111698133A (en) * | 2020-06-23 | 2020-09-22 | 上海仁童电子科技有限公司 | Waveform correction instrument and waveform correction method |
| CN111698133B (en) * | 2020-06-23 | 2021-12-24 | 上海仁童电子科技有限公司 | Waveform correction instrument and waveform correction method |
| CN112201263A (en) * | 2020-10-16 | 2021-01-08 | 广州资云科技有限公司 | Electric tone adjusting system based on song recognition |
| CN113470670A (en) * | 2021-06-30 | 2021-10-01 | 广州资云科技有限公司 | Method and system for quickly switching tone of electric tone |
| CN113470670B (en) * | 2021-06-30 | 2024-06-07 | 广州资云科技有限公司 | Method and system for rapidly switching electric tone basic tone |
| CN114449339A (en) * | 2022-02-16 | 2022-05-06 | 深圳万兴软件有限公司 | Background sound effect conversion method and device, computer equipment and storage medium |
| CN114449339B (en) * | 2022-02-16 | 2024-04-12 | 深圳万兴软件有限公司 | Background sound effect conversion method and device, computer equipment and storage medium |
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