CN101123088A - A chorus special effect processing method and system - Google Patents
A chorus special effect processing method and system Download PDFInfo
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Abstract
The invention discloses a special chorus effect processing method, which comprises: carrying out time-delaying for original signals through a delay time function to acquire at least one channel of signals comprising the chorus signals. Furthermore, the invention also discloses a special chorus effect processing system, which comprises at least one synthesize module and at least one function delay module. The synthesize module is used for overlaying the received channels of signals to synthesize chorus signal; each function delay module is used for carrying out time-delaying for the original signals through the delay time function to acquire one channel of signals, and outputting the channel of signals to the synthesizing module. The technical scheme disclosed in the invention can improve chorus effects.
Description
Technical Field
The invention relates to the technical field of sound signal processing, in particular to a chorus special effect processing method and system.
Background
The chorus special effect is an audio special effect technology, and aims to make a single voice processed by an algorithm increase the chorus so as to sound with the chorus effect. The principle of the conventional chorus special effect processing method is shown in fig. 1, which is to perform simple delay mixing processing on signals, generally, the original signal is delayed for a plurality of 35-50ms, and the original signal is combined with each delayed signal, so that one sound sounds like two or more sounds, and the transmission function of the sound is: y [ n ] = x [ n ] + x [ n-d1] + x [ n-d2] + … + x [ n-dm ], wherein y [ n ] is chorus signal, x [ n ] is original signal, x [ n-di ], di = d1, d 2. Although the effect of chorus of a plurality of voices can be generated, the effect of chorus is not obvious because the voices are basically the same because of simple reproduction of the original voice.
Disclosure of Invention
In view of the above, the present invention provides a chorus special effect processing method on one hand and a chorus special effect processing system on the other hand, so as to improve the chorus effect.
The chorus special effect processing method provided by the invention comprises the following steps:
and carrying out time delay processing on the original signal by adopting a delay time function to obtain at least one path of signal forming the chorus signal.
The time delay processing of the original signal by using the delay time function is as follows:
setting an independent variable as time and a dependent variable as a delay time function of a low-frequency signal which changes randomly along with the change of the time;
and taking the current function value of the delay time function as a delay factor, and performing delay processing on the original signal by using the delay factor.
Wherein the delay time function is: Δ d = [ Asin (n/FREQ) ], where n is the independent variable, Δ d is the dependent variable, FREQ is a preset delay change period, a is a preset delay amplitude, [ ] is a rounding symbol;
if the current function value as the delay factor is Δ dn, and
is an integer, the original signal is processed by the delay factorThe delay processing is as follows: for original signal x [ n ]]Performing delay processing to x [ n-delta dn]；
If the current function value as the delay factor is Δ dn, and
if the delay factor is a non-integer, the delaying the original signal by the delay factor is as follows: for original signal x [ n ]]Performing delay processing of
Wherein Δ d1 and Δ d2 are AND
Two adjacent integers.
Furthermore, the method further comprises: and carrying out tone modification processing on the original signal to obtain at least one path of signal forming the chorus signal.
Wherein, the tone-changing processing of the original signal comprises: according to the speed change proportion, calculating the cross-correlation value of the input signal and the output signal of the original signal corresponding to each search point in the cross-correlation search range, determining the maximum cross-correlation value in each stage, determining the final maximum cross-correlation value by combining the cross-correlation weight of each stage, and overlapping the input signal and the output signal of the original signal corresponding to the finally determined maximum cross-correlation value to obtain a signal after speed change;
and according to the tone-changing proportion, carrying out variable sampling rate processing on the signals after the speed change to obtain the signals after the tone changing.
Furthermore, the method further comprises: and performing sound change processing on the original signal to obtain at least one path of signal forming the chorus signal.
Wherein, the acoustic processing of the original signal comprises: carrying out tone-changing processing on an original signal to obtain a first signal; performing timbre processing on the first signal to obtain a second signal; and carrying out frequency spectrum equalization processing on the second signal to obtain a signal after sound change.
Wherein, the performing the tone processing on the first signal to obtain the second signal includes:
windowing the first signal to obtain a signal z (n);
converting the signal Z (n) from a time domain to a frequency domain to obtain a signal Z (k);
extracting a spectral envelope U (k) from said signal Z (k);
expanding or compressing the spectral envelope U (k) to obtain a new spectral envelope U' (k);
determining a weighting coefficient UO (k) of the signal Z (k) according to the spectrum envelopes U (k) and U '(k), and weighting the signal Z (k) to obtain a signal Z' (k);
converting the signal Z '(k) from a frequency domain to a time domain to obtain a signal Z' (n);
and windowing the signal z' (n) to obtain the second signal.
Preferably, before forming the chorus signal, further comprising: and weighting each path of signal.
The chorus special effect processing system provided by the invention comprises:
the synthesis module is used for synthesizing chorus signals after superposing the received signals;
and each function delay module is used for performing delay processing on the original signal by adopting a delay time function to obtain a path of signal and outputting the obtained path of signal to the synthesis module.
Wherein the function delay module comprises:
the delay factor calculation module is used for calculating a current function value according to the preset delay time function of the low-frequency signal with the independent variable as time and the dependent variable as time variation, and taking the calculated value as a delay factor;
and the delay processing module is used for performing delay processing on the original signal by using the delay factor obtained by the delay factor calculating module to obtain a path of signal and outputting the path of signal to the synthesizing module.
In addition, the system further comprises: at least one transposition module, wherein,
each tone modification module is used for performing tone modification processing on the original signal to obtain a path of signal, and outputting the obtained path of signal to the synthesis module.
Wherein, the tonal modification module comprises:
the variable speed processing module is used for calculating the cross-correlation value of the input signal and the output signal of the original signal corresponding to each search point in the cross-correlation search range according to the variable speed ratio, determining the maximum cross-correlation value in each stage, determining the final maximum cross-correlation value by combining the cross-correlation weight of each stage, and overlapping the input signal and the output signal of the original signal corresponding to the finally determined maximum cross-correlation value to obtain a variable speed signal;
and the tone-changing processing module is used for carrying out variable sampling rate processing on the signals after the speed change obtained by the speed-changing processing module according to the tone-changing proportion to obtain a path of signals after tone changing and outputting the path of signals to the synthesis module.
In addition, the system further comprises: at least one sound variation module, wherein,
each sound changing module is used for carrying out sound changing processing on the original signal to obtain a path of signal, and outputting the obtained path of signal to the synthesis module.
Wherein the sound changing module comprises:
the system comprises a tone-changing processing module, a first signal processing module and a second signal processing module, wherein the tone-changing processing module is used for carrying out tone-changing processing on an original signal to obtain a first signal;
the tone processing module is used for carrying out tone processing on the first signal to obtain a second signal;
and the frequency spectrum equalization processing module is used for carrying out frequency spectrum equalization processing on the second signal to obtain a path of signal after sound change and outputting the path of signal to the synthesis module.
Preferably, the system further comprises: and the coefficient weighting module is used for performing weighting processing on each path of signal before being input into the synthesis module and outputting the signal to the synthesis module.
According to the scheme, at least one path of signals forming the chorus signal is obtained by carrying out delay processing on the original signals by adopting the delay time function, so that improved delay mixed processing is carried out on the time domain, the synchronization of the sound is more natural, and the chorus effect is effectively improved.
In addition, the invention can also obtain at least one path of signals forming the chorus signal by carrying out tone modification processing on the original signal, thereby not only carrying out improved delay mixing processing on the time domain, but also carrying out tone modification processing on the frequency domain, leading the processed signal to have rich layering sense in the time domain and the frequency domain, and greatly improving the effect of chorus special effect.
Furthermore, at least one path of signals forming the chorus signal can be obtained by performing sound change processing on the original signal, so that on the basis of realizing voice sound change through tone change, the tone color adjustment and the frequency spectrum equalization processing are added according to the characteristics of the voice, the voice subjected to sound change processing is more natural, and the chorus effect is obviously improved.
Finally, the invention makes the strength of each path of signal in the chorus signal adjustable by weighting each path of signal before synthesizing the chorus signal, the chorus effect can be flexibly controlled according to the requirement, and the whole volume meets the requirement, and the chorus effect is optimal.
Drawings
Fig. 1 is a schematic diagram of a chorus special effect processing method in the prior art.
Fig. 2 is a schematic diagram of a chorus special effect processing method in an embodiment of the invention.
FIG. 3 is a flow chart of a tone processing method according to an embodiment of the present invention.
Fig. 4 is a schematic structural diagram of a chorus special effect processing system in the embodiment of the present invention.
Detailed Description
In the embodiment of the invention, at least one path of signal in the chorus signal is obtained by carrying out delay processing on the original signal by adopting a delay time function.
Further, at least one path of signal in the chorus signal is obtained by carrying out tone changing processing on the original signal.
Furthermore, at least one path of signal in the chorus signal is obtained by performing sound change processing on the original signal.
In order to make the objects, technical solutions and advantages of the present invention more apparent, the present invention is further described in detail below with reference to the following embodiments and the accompanying drawings.
Fig. 2 is a schematic diagram of a chorus special effect processing method in an embodiment of the invention. As shown by the solid line part in fig. 2, in this method, at least one path of signal constituting the chorus signal can be obtained by delaying the original signal by using a delay time function, and is denoted as x [ n- Δ d ]. Where Δ d is the time of the delay.
In a specific implementation, different delay time functions may be set for at least one of the signals forming the chorus signal, and the types of the delay time functions may be various, for example, one of the delay time functions may be: the independent variable is time, the dependent variable is delay time function of low frequency signal which changes randomly along with time change, when delay time function is adopted to delay each path of signal, the current function value of the delay time function corresponding to the current sampling point can be used as the current delay factor, and the delay factor is utilized to delay the original signal.
For example, for one of the signals, a delay time function shown in the following formula can be set:
Δ d = [ Asin (n/FREQ) ], where n is the independent variable time, Δ d is the dependent variable function value, FREQ is the preset delay variation period, a is the preset delay amplitude, and [ ] is the rounded symbol. Wherein, the values of FREQ and A can be determined according to the needs. For signals of different paths, the values of FREQ and A can be different.
For the time delay function, if the current function value as the delay factor is Δ dn, and
is an integer, when the original signal is delayed by the delay factor, the original signal x [ n ] can be directly delayed]The delay processing is carried out as follows: x [ n-Delta dn]。
If the current function value as the delay factor is Δ dn, and
if the delay factor is non-integer, then the delay factor is used to delay the original signal, so as to avoid x [ n- Δ dn) directly performing delay processing according to Δ dn due to the time variation of Δ n]The discontinuity of the representative signal, and thus the noise, can be smoothed by a smoothing method, i.e. by using a smoothing filter
Two adjacent integers delta d1 and delta d2 and a delay signal obtained by using the two integers delta d1 and delta d2 as delay factors are used for calculating x [ n-delta dn [ ]]The specific calculation process of the represented signal can be shown as follows:
that is, when the original signal is delayed by the delay factor, the original signal x [ n ] is delayed]The delay processing is carried out as follows:
the method can be as follows:
in addition, in a specific implementation, the time delay function may be other functions besides the above-described function, and may be specifically set according to actual needs.
As shown in the dotted line part in fig. 2, the method may further include: at least one path of signals forming the chorus signal is obtained by carrying out tone modification processing on an original signal and is recorded as w (n).
The purpose of pitch modification is to change the pitch of a sound, either up or down, without changing the overall length of the sound. The process of transposition is actually to spread the spectrum of the sound while the original relationship between the harmonic components is still preserved.
In the specific implementation, the pitch-changing processing method can be various, and is mainly divided into two categories, namely a time domain algorithm and a frequency domain algorithm. The frequency domain algorithm is generally implemented by a Phase Vocoder (Phase Vocoder); the time domain algorithm is generally realized by adopting a method of changing speed and changing sampling rate, and in addition, the time domain algorithm can also be realized by adopting an improved method of changing speed and changing sampling rate.
The improved method for adding the variable sampling rate to the variable speed mainly comprises the following steps: according to the speed change proportion, calculating the cross correlation value of the input signal and the output signal of the original signal corresponding to each search point in the cross correlation search range, determining the maximum cross correlation value in each stage, determining the final maximum cross correlation value by combining the cross correlation weight of each stage, and overlapping the input signal and the output signal of the original signal corresponding to the finally determined maximum cross correlation value to obtain a signal after speed change; and according to the tone-changing proportion, carrying out variable sampling rate processing on the signals after the speed change to obtain the signals after the tone changing.
Wherein, according to the proportion of changing tone, the signal after changing speed is processed with the variable sampling rate including: determining an up-sampling coefficient and a down-sampling coefficient according to the tone-changing proportion, dividing the up-sampling coefficient into more than one sub up-sampling coefficient, carrying out hierarchical up-sampling and low-pass filtering processing on the signals after speed changing according to each sub up-sampling coefficient, and then carrying out down-sampling processing on the signals after the low-pass filtering processing according to the down-sampling coefficient.
In addition, for a more detailed description of the improved method of variable speed plus variable sampling rate, please refer to the published invention patent of Chinese application number "200610152083.1".
As also shown in the dotted line part in fig. 2, the method may further include: at least one path of signals forming the chorus signal is obtained by performing sound variation processing on an original signal and is marked as v (n).
Wherein, the acoustic processing of the original signal mainly comprises: carrying out tone-changing processing on an original signal to obtain a first signal; performing timbre processing on the first signal to obtain a second signal; and carrying out frequency spectrum equalization processing on the second signal to obtain a signal v (n) after sound change.
The method of the pitch modulation processing may be the same as the described pitch modulation method, that is, the described time domain algorithm or frequency domain algorithm may be used.
The specific method for performing spectrum equalization processing on the speech signal may be: dividing the whole frequency spectrum into M blocks, weighting the spectral lines in each frequency spectrum block by using a preset coefficient in the direction of an amplitude axis, wherein the frequency spectrum obtained by weighting is the voice frequency spectrum subjected to equalization processing.
The entire spectrum is divided into M blocks, spectral lines are not evenly distributed, spectral lines in the low-frequency region include a low spectral line interval density, and spectral lines in the high-frequency region include a high spectral line interval density. For example, 0HZ to 10HZ may be divided into one block, and 500HZ to 1000HZ may be divided into one block.
Since the spectral equalization process changes the amplitude value in a certain region in the speech spectrum, the effect of changing the tone color can also be achieved.
The above method of performing spectrum equalization processing on a signal is a common spectrum equalization processing method, which is well known to those skilled in the art, and therefore, is not described herein in detail.
Fig. 3 is a flowchart of performing timbre processing on a first signal in the embodiment of the present invention, and as shown in fig. 3, the specific steps include:
In the case of spectral analysis of a signal, an infinitely long signal is analyzed. This is practically impossible and only a limited length of signal analysis can be performed. The interception process is to multiply the infinite signal by a window function with a finite width and to perform power spectrum analysis on the intercepted signal to obtain an approximate spectrum. This process is a windowing of the signal. In the present embodiment, the window function selects a sine window.
Before windowing the signal, it is necessary to synthesize the signal w (N) with length N and the last input signal w _ old (N) with length N in the previous frame signal into a large frame with length 2N
Wherein n represents time and n > 0.
Then, conducting sine window processing on the synthesized large frame w' (n) to obtain a signal frame z (n):
z (N) = w' (N) sin (N/2N), N =0,1,2.. 2N-1,n represents time.
The spectral envelope is a curve used to represent the general trend of the spectrum in a two-dimensional spectrogram. There are many methods for extracting the spectral envelope from a spectrogram, and among them, the more common methods are windowing smoothing and median smoothing.
When a windowing smoothing method is utilized, a window function W (k) with the window length of 2L +1 is adopted, the spectral line amplitude values in the window are averaged, the average value is taken as the value of a longitudinal coordinate, the value of the transverse central point of the window is taken as the value of an abscissa, and each point obtained in the way is connected by a smoothing curve to obtain a spectral envelope U (k):
when a median smoothing method is utilized, a window function W (k) with the window length of 2L +1 is adopted, the amplitude value corresponding to the frequency spectrum at the middle position in each window is taken as the value of the ordinate, the value of the transverse center point of the window is taken as the value of the abscissa, and each obtained point is connected by a smoothing curve to obtain a spectrum envelope U (k):
U(k)＝Mid{|Z(k+i)|}，i＝-L～L。
in this embodiment, the window functions in the above two methods may both adopt rectangular windows.
For the value of L, if L takes 0, the expression of the window function is 1, which is equivalent to processing each spectral line, and the significance of the segmented processing is lost; if L takes too large a value, it may result in the envelope curve being too smooth, which is far from the actual situation. Therefore, in this embodiment, L may take 2.
The adjustment of the spectral envelope mainly comprises the transverse expansion or compression of the spectral envelope, so that the effect of changing the position of a resonance peak is achieved. The expansion or compression of the spectral envelope is obtained by multiplying the frequency value by a coefficient beta, and the adjusted spectral envelope expression is as follows: u' (k) = U (β k). As can be seen from the expression, when β is greater than 1, the spectral envelope is compressed; when β is less than 1, the spectral envelope is expanded; beta is equal to 1, the spectral envelope is unchanged.
Firstly, the final spectral envelope weighting coefficient UO (k) is obtained by dividing the spectral envelope functions before and after adjustment,
then, uo (k) weighting is carried out on the signal Z (k) to obtain a signal Z' (k), and the spectral envelope shaping of the voice signal is realized.
The expression of the shaped voice signal is as follows: z' (k) = Z (k) Uo (k).
The purpose of windowing the signal is the same as that in step 301, and in order to intercept an infinite-length signal, the power spectrum analysis is performed on the intercepted signal to obtain an approximate spectrum. In this embodiment, the window function still selects the sine window.
After windowing, the signal z ω (n) is obtained: z ω (N) = z' (N) sin (N/2N), N =0,1,2.
Considering that when the signal is processed frame by frame, there is a possibility that the signal output is discontinuous after the signal is processed by the sound changing processing. Therefore, the first half signal of the current window can be superimposed with the result saved in the previous window to obtain the final second signal c (n):
c (N) = z ω (N) + z ω '(N), N =0,1,2.. N-1, where z ω' (N) represents the result saved in the last frame.
The first half of the windowed signal is overlapped with the result stored in the previous window, and the second half of the windowed result is stored for being overlapped with the first half of the windowed signal in the next window, so that the voice signal after changing voice can be continuously output.
The sound changing method in the above process is based on changing tone to realize voice sound changing, and according to the characteristics of sound, the sound color adjustment and the frequency spectrum equalization are added, so that the sound processed by sound changing is more natural.
Finally, as shown in fig. 2, after the signals are superimposed, a chorus signal is formed, and in order to adjust the sound level of the chorus signal and meet the acquisition requirement of the signals, the signals before the chorus signal is formed can be further weighted by using a preset weighting coefficient, and then the final chorus signal is synthesized. Such as alpha, beta, and gamma in fig. 2.
In the above method, in addition to the signals after the delay processing by using the delay time function, the modified signals, and the like described above, there may be other signals, such as the original signals, the signals delayed according to the prior art, or the signals processed in other manners.
The chorus special effect processing method in the embodiment of the present invention is described in detail above, and the chorus special effect processing system in the embodiment of the present invention is described in detail below.
Fig. 4 is a schematic structural diagram of a chorus special effect processing system in the embodiment of the present invention. As shown in the solid line portion in fig. 4, the system includes at least: a synthesis module and at least one delay module. In addition, there may be other processing modules, such as a delay processing module in the prior art, or a processing module adopting other manners.
The synthesis module is used for synthesizing the chorus signal after superposing the received signals.
Each delay module is used for carrying out delay processing on the original signal by adopting a delay time function to obtain a path of signal, and outputting the obtained path of signal to the synthesis module.
In specific implementation, the delay module may have multiple implementation forms, and one of the implementation forms may specifically include: a delay factor calculating module and a delay processing module.
The delay factor calculation module is used for calculating a current function value according to a preset delay time function of a low-frequency signal with an independent variable as time and a dependent variable as a random change along with the time, and the calculated function value is used as a delay factor.
And the delay processing module is used for delaying the original signal by using the delay factor obtained by the delay factor calculating module and outputting the delayed original signal to the synthesizing module.
The specific operation process of the delay module and its internal modules may be consistent with the specific operation process described in the method shown in fig. 2. Wherein the delay time function may also correspond to the delay time function described in the method of fig. 2.
As shown in phantom in fig. 4, the system may further comprise: at least one transposition module.
Each tone modification module is used for performing tone modification processing on the original signal to obtain a path of signal, and outputting the obtained path of signal to the synthesis module.
In specific implementation, the tone modification module may be implemented in various ways, and the specific operation process thereof may be consistent with the specific operation process described in the method illustrated in fig. 2. Corresponding to the improved method for adding the variable sampling rate by the variable speed, the tone varying module may specifically include: the device comprises a variable speed processing module and a tone changing processing module.
The variable speed processing module is used for calculating the cross-correlation value of the input signal and the output signal of the original signal corresponding to each search point in the cross-correlation search range according to the variable speed ratio, determining the maximum cross-correlation value in each stage, determining the final maximum cross-correlation value by combining the cross-correlation weight of each stage, and overlapping the input signal and the output signal of the original signal corresponding to the finally determined maximum cross-correlation value to obtain a signal after variable speed;
and the tone-changing processing module is used for carrying out variable sampling rate processing on the signals after the speed change obtained by the speed-changing processing module according to the tone-changing proportion to obtain a path of signals after tone changing and then outputting the path of signals to the synthesis module.
The specific operation process and implementation of each module can be consistent with the specific operation process described in the method shown in fig. 2, and refer to the published patent application No. 200610152083.1 of the invention.
In addition, the system may further include: at least one sound changing module.
Each of the sound changing modules is used for performing sound changing processing on the original signal to obtain a path of signal, and outputting the obtained path of signal to the synthesis module.
When the sound changing module is specifically implemented, the sound changing module can have multiple implementation forms, and one of the implementation forms can include: the system comprises a tone-changing processing module, a tone processing module and a frequency spectrum balancing processing module.
The tone-changing processing module is used for carrying out tone-changing processing on the original signal to obtain a first signal.
The specific operation process of the transposition processing module can be consistent with the specific operation process of the transposition module.
And the tone processing module is used for carrying out tone processing on the first signal to obtain a second signal.
And the frequency spectrum equalization processing module is used for carrying out frequency spectrum equalization processing on the second signal to obtain a path of sound-changed signal and then outputting the path of sound-changed signal to the synthesis module.
When the tone color processing module is specifically implemented, the tone color processing module may include: the device comprises a first windowing processing module, a first conversion module, an envelope extraction module, a new envelope acquisition module, a weighting module, a second conversion module and a second windowing processing module.
The first windowing processing module is used for carrying out windowing processing on the first signal to obtain a signal z (n).
The first conversion module is configured to convert the signal Z (n) from the time domain to the frequency domain to obtain a signal Z (k).
The envelope extraction module is used for extracting a spectral envelope U (k) from the signal Z (k).
And the new envelope acquisition module is used for expanding or compressing the spectrum envelope U (k) to obtain a new spectrum envelope U' (k).
The weighting module is used for determining a weighting coefficient UO (k) of the signal Z (k) according to the spectrum envelopes U (k) and U '(k), and weighting the signal Z (k) to obtain a signal Z' (k).
The second conversion module is configured to convert the signal Z '(k) from the frequency domain to the time domain to obtain a signal Z' (n).
The second windowing processing module is used for windowing the signal z' (n) to obtain the second signal.
Finally, the system may further comprise: and the coefficient weighting module is used for carrying out weighting processing on each path of signals before being input into the synthesis module and then outputting the signals to the synthesis module.
The above-mentioned embodiments are intended to illustrate the objects, technical solutions and advantages of the present invention in further detail, and it should be understood that the above-mentioned embodiments are only preferred embodiments of the present invention, and are not intended to limit the scope of the present invention, and any modifications, equivalents, improvements and the like made within the spirit and principle of the present invention should be included in the scope of the present invention.
Claims (16)
1. A chorus special effect processing method is characterized by comprising the following steps:
and carrying out delay processing on the original signal by adopting a delay time function to obtain at least one path of signal forming the chorus signal.
2. The method of claim 1, wherein delaying the original signal with the delay time function is:
setting an independent variable as a time function and a dependent variable as a delay time function of a low-frequency signal which randomly changes along with the time change:
and taking the current function value of the delay time function as a delay factor, and performing delay processing on the original signal by using the delay factor.
3. The method of claim 2, wherein the delay time function is: Δ d = [ Asin (n/FREQ) ], wherein n is the independent variable, Δ d is the dependent variable, FREQ is a preset delay change period, a is a preset delay amplitude, and [ ] is a rounded symbol;
if the current function value as the delay factor is Δ dn, andis an integer, thenThe delay processing of the original signal by using the delay factor comprises the following steps: the original signal x [ n ]]Performing delay processing to x [ n-delta dh]；
If the current function value as the delay factor is Δ dn, and
if the delay factor is a non-integer, the delaying the original signal by the delay factor is as follows: for original signal x [ n ]]Performing a delay process ofWherein Ad1 and Ad2 are andtwo adjacent integers.
4. The method of claim 1, further comprising: and carrying out tone-changing processing on the original signal to obtain at least one path of signal forming the chorus signal.
5. The method of claim 4, wherein the pitch-shifting the original signal is: according to the speed change proportion, calculating the cross-correlation value of the input signal and the output signal of the original signal corresponding to each search point in the cross-correlation search range, determining the maximum cross-correlation value in each stage, determining the final maximum cross-correlation value by combining the cross-correlation weight of each stage, and overlapping the input signal and the output signal of the original signal corresponding to the finally determined maximum cross-correlation value to obtain a signal after speed change;
and according to the tone-changing proportion, carrying out variable sampling rate processing on the signals after the speed change to obtain the signals after the tone changing.
6. The method of claim 1, further comprising: and performing sound change processing on the original signal to obtain at least one path of signal forming the chorus signal.
7. The method of claim 6, wherein the voicing the raw signal is: carrying out tone-changing processing on an original signal to obtain a first signal; performing timbre processing on the first signal to obtain a second signal; and carrying out frequency spectrum equalization processing on the second signal to obtain a signal after changing the sound.
8. The method of claim 7, wherein performing timbre processing on the first signal to obtain the second signal comprises:
windowing the first signal to obtain a signal z (n);
converting the signal Z (n) from a time domain to a frequency domain to obtain a signal Z (k);
extracting a spectral envelope U (k) from said signal Z (k);
expanding or compressing the spectral envelope U (k) to obtain a new spectral envelope U' (k);
determining a weighting coefficient UO (k) of the signal Z (k) according to the spectrum envelopes U (k) and U '(k), and weighting the signal Z (k) to obtain a signal Z' (k);
converting the signal Z '(k) from a frequency domain to a time domain to obtain a signal Z' (n);
and windowing the signal z' (n) to obtain the second signal.
9. The method of any of claims 1 to 8, prior to forming the chorus signal, further comprising: and weighting each path of signal.
10. A chorus special effects processing system, the system comprising:
the synthesis module is used for synthesizing chorus signals after superposing the received signals;
characterized in that, this system still includes:
and each function delay module is used for performing delay processing on the original signal by adopting a delay time function to obtain a path of signal and outputting the obtained path of signal to the synthesis module.
11. The system of claim 10, wherein the functional delay module comprises:
the delay factor calculation module is used for calculating a current function value according to the preset delay time function of the low-frequency signal with the independent variable as time and the dependent variable as time variation, and taking the calculated value as a delay factor;
and the delay processing module is used for performing delay processing on the original signal by using the delay factor obtained by the delay factor calculating module to obtain a path of signal and outputting the path of signal to the synthesizing module.
12. The system of claim 10, further comprising: at least one tone modifying module, wherein,
each tone modification module is used for performing tone modification processing on the original signal to obtain a path of signal, and outputting the obtained path of signal to the synthesis module.
13. The system of claim 12, wherein the transposition module comprises:
the variable speed processing module is used for calculating the cross-correlation value of the input signal and the output signal of the original signal corresponding to each search point in the cross-correlation search range according to the variable speed ratio, determining the maximum cross-correlation value in each stage, determining the final maximum cross-correlation value by combining the cross-correlation weight of each stage, and overlapping the input signal and the output signal of the original signal corresponding to the finally determined maximum cross-correlation value to obtain a variable speed signal;
and the tone-changing processing module is used for carrying out variable sampling rate processing on the signals after the speed change obtained by the speed-changing processing module according to the tone-changing proportion to obtain a path of signals after tone changing and outputting the path of signals to the synthesis module.
14. The system of claim 10, further comprising: at least one sound changing module, wherein,
each sound changing module is used for carrying out sound changing processing on the original signal to obtain a path of signal and outputting the obtained path of signal to the synthesis module.
15. The system of claim 14, wherein the voicing module comprises:
the system comprises a tone-changing processing module, a first signal processing module and a second signal processing module, wherein the tone-changing processing module is used for carrying out tone-changing processing on an original signal to obtain a first signal;
the tone processing module is used for carrying out tone processing on the first signal to obtain a second signal;
and the frequency spectrum equalization processing module is used for carrying out frequency spectrum equalization processing on the second signal to obtain a path of signal after sound change and outputting the path of signal to the synthesis module.
16. The system of any one of claims 10 to 15, further comprising: and the coefficient weighting module is used for performing weighting processing on each path of signal before being input into the synthesis module and outputting the signal to the synthesis module.
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| CN2007101213039A CN101123088B (en) | 2007-09-03 | 2007-09-03 | A chorus special effect processing method and system |
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Cited By (2)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN102436805A (en) * | 2010-09-29 | 2012-05-02 | 炬力集成电路设计有限公司 | Reverberant unit and reverberating method |
| WO2017125840A1 (en) * | 2016-01-19 | 2017-07-27 | Hua Kanru | Method for analysis and synthesis of aperiodic signals |
Family Cites Families (4)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| US4080861A (en) * | 1976-07-09 | 1978-03-28 | Thomas International Corporation | Chorus control for electronic musical instrument |
| US5444180A (en) * | 1992-06-25 | 1995-08-22 | Kabushiki Kaisha Kawai Gakki Seisakusho | Sound effect-creating device |
| JP3173382B2 (en) * | 1996-08-06 | 2001-06-04 | ヤマハ株式会社 | Music control device, karaoke device, music information supply and reproduction method, music information supply device, and music reproduction device |
| CN100375149C (en) * | 2005-08-19 | 2008-03-12 | 北京中星微电子有限公司 | A reverberation generating circuit |
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- 2007-09-03 CN CN2007101213039A patent/CN101123088B/en not_active Expired - Fee Related
Cited By (3)
| Publication number | Priority date | Publication date | Assignee | Title |
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| CN102436805A (en) * | 2010-09-29 | 2012-05-02 | 炬力集成电路设计有限公司 | Reverberant unit and reverberating method |
| CN102436805B (en) * | 2010-09-29 | 2013-03-27 | 炬力集成电路设计有限公司 | Reverberant unit and reverberating method |
| WO2017125840A1 (en) * | 2016-01-19 | 2017-07-27 | Hua Kanru | Method for analysis and synthesis of aperiodic signals |
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