JP3984021B2 - Speech / acoustic signal encoding method and electronic apparatus - Google Patents

Description

[0001]
BACKGROUND OF THE INVENTION
The present invention relates to an audio / acoustic signal encoding method and an electronic apparatus.
[0002]
[Prior art]
A CELP (Code-Excited Linear Prediction) system is known as a method for compressing and encoding an audio signal [“Code-Excited Linear Prediction (CELP): High-quality Speech at Very Low Rates” Proc.ICASSP'85, 25 1.1. pp. 937-940, 1985].
[0003]
In the CELP method, a speech signal is modeled by dividing it into a synthesis filter and a sound source signal for driving the synthesis filter. The encoded synthesized speech signal is generated by passing the sound source signal through a synthesis filter.
[0004]
The excitation signal is generated by combining two code vectors of an adaptive code vector generated from an adaptive code book that stores past excitation signals and a noise code vector generated from a noise code book.
[0005]
The adaptive code vector has a role of representing a repetition of a waveform with a pitch period, which is a characteristic of a sound source signal in a voiced sound section. On the other hand, the noise code vector has a role of supplementing components included in the sound source signal that cannot be represented by the adaptive code vector, and is used to make the synthesized speech signal more natural.
[0006]
The CELP system is characterized in that encoding of a sound source signal makes it difficult to perceive encoding distortion by evaluating distortion at the level of an audio signal weighted by auditory weight. The reason why the encoding distortion becomes difficult to perceive is because auditory weighting is performed so that the spectrum of the encoding distortion is masked in the shape of the spectrum of the audio signal, and frequency masking is used. The auditory weight characteristic in this case is obtained from the speech signal for each encoding section, and the sound source signal is encoded using the same auditory weight characteristic in the same encoding section.
[0007]
Here, when the encoding bit rate is reduced to, for example, about 4 kbit / s in the case of an audio signal, the number of bits allocated to express the sound source signal is insufficient, so that distortion due to encoding is perceived as sound. become. As a result, the deterioration of sound quality such as sound fading or noise mixing becomes remarkable.
[0008]
For this reason, there is a need for highly efficient encoding that can generate high-quality synthesized speech even when the bit rate is reduced. Such a requirement applies to the encoding of an acoustic signal.
[0009]
[Problems to be solved by the invention]
As described above, in the conventional speech / acoustic signal encoding method, the auditory weight characteristic is obtained from the speech signal for each encoding section, and the sound source signal is encoded using the same auditory weight characteristic in the encoding section. Therefore, there is a problem that it is difficult to obtain high-quality synthesized speech at a low bit rate.
[0010]
The present invention has been made in view of the above points, and it is an object of the present invention to provide a speech / acoustic signal encoding method and an electronic apparatus that can generate a high-quality speech signal / acoustic signal even at a low bit rate.
[0011]
[Means for Solving the Problems]
In order to achieve the above object, a first aspect of the present invention is an audio / acoustic signal encoding method for generating a synthesized sound signal by passing a sound source signal through a synthesis filter, the input signal For the time series signal corresponding to the component that could not be predicted, , To reduce distortion between synthesized sound signal and target sound signal A weight information acquisition step for acquiring position weight information, an evaluation step for evaluating the distortion of the synthesized sound signal and the target sound signal by weighting using the weight information acquired in the weight information acquisition step, and the evaluation A code selection step for performing code selection of parameters of the sound source signal so that distortion between the synthesized sound signal and the target sound signal is reduced based on the evaluation result of the step;
It comprises.
[0012]
According to a second aspect of the present invention, there is provided an audio / acoustic signal encoding method for generating a synthesized sound signal by passing a sound source signal through a synthesis filter. A time series signal corresponding to a component that could not be predicted for Amplitude or amplitude of time series signal Representing the size of A first evaluation step for evaluating a function value and a large amplitude or function value by the first evaluation step To reduce distortion at the sample position A weight information acquisition step for acquiring position weight information, and a second evaluation step for weighting and evaluating distortion between the synthesized sound signal and the target sound signal using the position weight information acquired in the weight information acquisition step. And a code selection step of selecting a code of a parameter of the sound source signal so as to reduce distortion between the synthesized sound signal and the target sound signal based on the evaluation result in the second evaluation step.
[0013]
According to a third aspect of the present invention, there is provided an audio / acoustic signal encoding method for generating a synthesized sound signal by passing a sound source signal through a synthesis filter. A time series signal corresponding to a component that could not be predicted for Time-series signal amplitude or Represents the magnitude of the amplitude Evaluate the function value and with respect to the sample position where the amplitude or function value is large Corresponds to the weight value The first position weight information is obtained, and the first position weight information is obtained with respect to the sample position having a small amplitude or function value. Corresponds to a small weight value different from A weight information acquisition step for acquiring second position weight information, and a distortion between the synthesized sound signal and the target sound signal are weighted using the first and second weight information acquired in the weight information acquisition step. And a code selection step for selecting a code of a parameter of the sound source signal so that distortion between the synthesized sound signal and the target sound signal is reduced based on the evaluation result of the evaluation step. To do.
[0014]
According to a fourth aspect of the present invention, in any one of the first to third aspects, the parameter of the excitation signal searches for an adaptive codebook or a noise codebook used for obtaining the excitation signal. It is information for.
[0015]
Further, the fifth aspect of the present invention is the first to the first 4 In any one of the aspects, the time-series signal obtained from the input signal is either a prediction residual signal or a simulation signal of the prediction residual signal.
[0016]
According to a sixth aspect of the present invention, in any one of the first to fifth aspects, the parameter of the sound source signal is selected using the position weight information and the auditory weight.
[0017]
According to a seventh aspect of the present invention, there is provided an electronic device, comprising: an input unit for inputting a voice / acoustic signal; and an encoding process performed on the voice / acoustic signal input via the input unit. An encoding unit that performs the transmission, a transmission unit that transmits the audio / acoustic signal encoded by the encoding unit, a reception unit that receives the encoded audio / acoustic signal, and the reception unit received via the reception unit A decoding unit that performs a decoding process on the audio / acoustic signal; and an output unit that outputs the audio / acoustic signal decoded by the decoding unit. The encoding unit includes: The encoding method according to any one of the sixth to sixth aspects is executed.
[0018]
DETAILED DESCRIPTION OF THE INVENTION
Hereinafter, an embodiment of the present invention will be described with reference to the drawings.
[0019]
FIG. 1 is a block diagram showing the main structure of the speech / acoustic signal encoding method of the present invention. Here, an example in which the present invention is applied to CELP coding of an audio signal will be described.
[0020]
A voice input from a voice input means (not shown) such as a microphone is subjected to A / D conversion and is input as a discrete voice signal from the input terminal 100 at predetermined time intervals. Usually, this time interval has a length of about 10 to 30 ms and is sometimes called a frame length.
[0021]
In the CELP system, the vocal cord signal is made to correspond to the sound source signal as a model of the speech generation process, the spectral envelope characteristic represented by the vocal tract is represented by the synthesis filter, the sound source signal is input to the synthesis filter, and the speech signal is expressed by the output of the synthesis filter To do. The present invention is the same as the CELP method in that the sound source signal is encoded so that the waveform distortion between the input voice signal and the synthesized voice signal is aurally small, but the calculation of the waveform distortion used in the codebook search is performed. The point which introduces position weight is different from the past.
[0022]
That is, the CELP coding according to the present invention described here is not limited to the spectrum parameter codebook search unit 911, the adaptive codebook search unit 912, the noise codebook search unit 913, and the gain codebook search unit 914, but also calculates a residual signal. Coding is performed using the unit 120 and the position weight control unit 910. The index information searched by each codebook search unit is output from the encoded data output unit 915 as audio encoded data.
[0023]
The functions of the individual codebook search units in the speech encoding shown in FIG. 1 will be described below.
[0024]
The spectrum parameter codebook search unit 911 inputs a speech signal from the input terminal 100 for each frame, searches a spectrum parameter codebook prepared in advance, and better expresses the spectrum envelope of the input speech signal. A possible codebook index (spectrum parameter code) A is selected, and this index is output to the encoded data output unit 915. Usually, in the CELP method, an LSP (Line Spectrum Pair) parameter is used as a spectrum parameter used for encoding a spectrum envelope. However, the present invention is not limited to this, and other parameters can be used as long as they can express the spectrum envelope. It is.
[0025]
The residual signal calculation unit 120 calculates a residual signal using the speech signal and the spectral parameters from the spectral parameter codebook search unit 911. As a specific example, a prediction residual signal r (n) is obtained by converting a spectral parameter into an LPC coefficient and filtering a speech signal with a prediction filter A (z) using the LPC coefficient. Since the detailed method of obtaining the prediction residual signal r (n) is known, the description thereof is omitted here. The prediction residual signal is sometimes called a residual signal. In the following description, it will be called a residual signal.
[0026]
The position weight control unit 910 obtains a position weight based on the residual signal obtained from the speech signal, and outputs the position weight to the adaptive codebook search unit 912, the noise codebook search unit 913, and the gain codebook search unit 914, respectively. At the same time, the search units 912, 913, and 914 of each codebook are controlled so that the position weight is reflected in the distortion evaluation value in each codebook.
[0027]
2 and 3 are diagrams for explaining the procedure for obtaining the position weight from the audio signal by the method of the present embodiment.
[0028]
FIG. 2A shows an example of a discrete waveform of a speech signal before encoding. In the figure, the waveform amplitude of the audio signal at the sample position n = i is represented as s (i). FIG. 2B is an example of a waveform of the residual signal obtained from the audio signal of FIG. Since the residual signal is an error signal when a speech signal is predicted, it can be said that a position where the amplitude of the residual signal is larger than the others is a position that could not be expressed sufficiently by prediction. The position is considered to be a position that includes more speech features that cannot be expressed by prediction than other positions with small amplitudes. Therefore, by introducing a mechanism for encoding a position where the amplitude of the residual signal is larger than other positions more accurately than other positions (that is, with less distortion) in encoding the sound source signal, a higher quality synthesized speech can be obtained. It becomes possible to provide.
[0029]
In the present invention, by analyzing the characteristics of the residual signal based on the residual signal, it is analyzed at which position the distortion should be reduced, and the position weight is set so that the distortion evaluation penalty becomes large at such a position. Set relatively large.
[0030]
Here, an example of a method for setting the position weight from the residual signal will be described with reference to FIG. In the figure, the simplest method for comparing the absolute value amplitude at each position of the residual signal with the threshold value 49 determined by a predetermined method and setting the position weight based on the magnitude relationship is shown. That is, if the absolute value amplitude of the residual signal at each position is smaller than the threshold value 49, the position weight is set relatively small. Conversely, if the absolute value amplitude is larger than the threshold value 49, the position weight is set. Is set relatively large. In fact, in the example of FIG. 2C, since the absolute value amplitude indicated by 50 is smaller than the threshold value 49, the position weight at this position is set relatively small, and the absolute value amplitude indicated by 51 is the threshold value 49. Therefore, the position weight at this position is set to be relatively large.
[0031]
The threshold value can be determined based on, for example, the square root average of the residual signal or the average absolute value of the residual signal. If the position weight is set using the normalized amplitude of the residual signal, the threshold value can be set to a substantially fixed value, but is not limited thereto.
[0032]
FIG. 3A shows an example of the position weight v (n) obtained as a result. In this example, the position weights v (n) all have the same polarity (all positive in this figure: v (n)> 0). This indicates that the position weight is a weight function determined with respect to the sample position n. Since the sample position n indicates the position n of the sampled time series signal, the position n referred to in the present invention may be considered as time n or time n. Therefore, it can be said that the weight v (n) related to the position is a position weight related to the sample position in the target coding section, and is a time weight (or time weight) related to the time n defined in this section. I can say that. Such weighting relating to the time position is a weighting defined so as to be multiplied for each sample of the time series signal, and is completely different from the weighting realized by the filter operation or the convolution operation used in the conventional auditory weighting. It is.
[0033]
FIG. 3B is an example in which a method of setting the position weight at a position where the absolute value amplitude of the residual signal is very small is adopted to a smaller value, and the size of the position weight is set to three types. For example, in the figure, the position weight v (21) is smaller than the value of v (21) in FIG. 3A because the absolute value amplitude of the residual signal at position n = 21 is very large. Reflects the smallness.
[0034]
As another method of setting the position weight, the position weight v (n) can be obtained by quantizing the absolute value using the residual signal r (n) or a signal obtained by normalizing the residual signal. . That is, v (n) = Q [abs (r (n))]. Here, abs () represents a function representing an absolute value, and Q [x] represents a quantized output when x is input to a predetermined quantizer Q. When the quantizer output is configured to use a binary quantizer, position weights of two kinds of sizes can be set as in FIG. Similarly, in the case of using a quantizer with a three-value quantization output, three types of position weights can be set in the same manner as in FIG. There may be four or more types of position weight magnitude settings.
[0035]
As another method for setting the position weight, the square signal {r (n)} of the residual signal is used instead of r (n). ² It is also possible to set the position weight by using the method shown in the above example.
[0036]
As described above, there are various methods for setting the position weight. In short, it is only necessary to have a mechanism that can reflect the importance of each position in the position weight. Even if it exists, it is included in this invention.
[0037]
Here, returning to FIG. 1, the description will be continued.
[0038]
The adaptive codebook search unit 912 is used to represent a component that repeats at a pitch period in the sound source signal. In the CELP method, the encoded past sound source signal is stored as an adaptive codebook for a predetermined length, and this is held in both the speech encoding unit and the speech decoding unit, thereby supporting the specified pitch period. Thus, the signal can be extracted from the adaptive codebook. In the adaptive codebook, since the output signal from the codebook and the pitch period correspond one to one, the pitch period can correspond to the index of the adaptive codebook.
[0039]
Under such a structure, adaptive codebook search section 912 uses distortion from position weight control section 910 as the distortion between the synthesized signal and the target speech signal when the output signal from the codebook is synthesized by the synthesis filter. Evaluation is performed at a level weighted by a weight, and a pitch period is searched so that the distortion is reduced. Then, the searched index (adaptive code) L is output to the encoded data output unit 915. The above weighting has an effect of selecting a code that is more effective in preventing distortion by using both the position weight and the conventional auditory weight.
[0040]
The noise codebook search unit 913 is used to represent a noisy component in the sound source signal. In the CELP system, the noise component of the sound source signal is expressed using a noise codebook. The structure is such that a noisy signal or a pulse-like signal can be extracted from the noise codebook corresponding to the specified index.
[0041]
In the present embodiment, it is written as a noise codebook, but it goes without saying that the noise signal represented by this codebook is not necessarily so-called noise. For example, the noise codebook may be a codebook that generates a pulsed excitation signal such as an algebraic codebook. The algebraic codebook is a codebook in which the amplitude of a predetermined number of pulses is limited to +1 and −1 and a code vector is represented by a combination of pulse position information and polarity information.
[0042]
As a feature of the algebraic codebook, since it is not necessary to store the code vector itself directly, the memory amount representing the codebook is small, and the calculation amount for selecting the code vector is relatively small. The noise component contained in sound source information can be expressed with high quality. As described above, what uses an algebraic codebook for encoding a sound source signal is called an ACELP system or an ACELP-based system, and it is known that synthesized speech with relatively little distortion can be obtained.
[0043]
Under such a structure, the noise codebook search unit 913 calculates the distortion between the synthesized speech signal reproduced using the output signal from the codebook and the target speech signal in the noise codebook search unit 913 by using position weights. Evaluation is performed at a level weighted by the position weight from the control unit 910, and an index (noise code) C that reduces the distortion is searched.
[0044]
One example of a method for selecting a noise codebook index (noise code) using the position weight v (n) is to select the index k of the noise code vector ck that minimizes the following distortion Dv.
[0045]
Dv = Σ [v (n) {X (n) −gHck (n)}] ² (1)
Here, X (n) is a target signal, g is a gain, H is an impulse response matrix for generating a synthesized speech signal, and ck (n) is an element at position n of the noise code vector. If defined in this way, the synthesized speech signal to be reproduced can be represented by gHck (n). In the conventional method, the index k of the noise code vector ck is selected so that the square sum of the error signal {X (n) -gHck (n)} between the target signal and the synthesized speech signal to be reproduced is minimized. Based on this, code selection is performed.
[0046]
Here, an error with a position weight (position weight) obtained by multiplying a position weight v (n) for each position n of the error signal {X (n) −gHck (n)} between the target signal and the synthesized speech signal to be reproduced. The index k of the random code vector ck is selected so that the sum of squares of (additional distortion) v (n) {X (n) −gHck (n)} is minimized. At this time, the position weight to be used may be obtained from the residual signal, but the position weight obtained from the audio signal or the audio signal subjected to auditory weighting can also be used.
[0047]
Instead of the residual signal, a simulated signal having a shape relatively close to the residual signal can be used. As such a residual signal simulation signal, for example, an adaptive code vector is conceivable, and it is also effective to obtain a position weight by using the adaptive code vector instead of the residual signal.
[0048]
The searched noise code C is output to the encoded data output unit 915. The above weighting has an effect of selecting a code in which distortion is less likely to be heard more effectively by combining position weighting and conventional auditory weighting.
[0049]
Next, the gain codebook search unit 914 is used to express the gain component of the excitation signal. In a typical CELP system, the gain codebook search unit 914 encodes two types of gains, a gain used for pitch components and a gain used for noise components. In the codebook search, the distortion between the synthesized speech signal reproduced using the gain candidates extracted from the codebook and the target speech signal is evaluated at a level weighted by the position weight from the position weight control unit 910, An index (gain code) G that reduces the distortion is searched. The searched gain code G is output to the encoded data output unit 915. The above weighting has an effect of selecting a code that is more effective in preventing distortion by using both the position weight and the conventional auditory weight.
[0050]
The encoded data unit 915 outputs the encoded data as audio encoded data.
[0051]
Here, the method of using position weights for each of the three codebook searches of adaptive codebook search, noise codebook search, and gain codebook search has been described. However, the present invention is not limited to this, and various modifications are possible. It goes without saying that examples are possible. For example, a method using position weights only for noise codebook search is also effective.
[0052]
This is the end of the description of speech encoding in FIG.
[0053]
FIG. 4 is a flowchart for explaining an encoding method according to an embodiment of the present invention.
[0054]
A speech signal is input for each predetermined coding section (step S1), a spectrum parameter codebook search is performed (step S2), and a residual signal is obtained from the speech signal (step S3). Next, the position weight v (n) of each position n is set according to the relative magnitude relationship of the obtained amplitude values r (n) of the residual signal (step S4). Then, an adaptive codebook search is performed using the position weight v (n) (step S5), and then a noise (algebraic) codebook search is performed using the position weight v (n) (step S6). A gain codebook search is performed using v (n) (step S7).
[0055]
Finally, the codes A, L, C, and G obtained by the above search are output (step S8). Then, it is determined whether or not encoding of the next encoding section is necessary (step S9). If it is not necessary, the encoding process is terminated.
[0056]
An example of a specific method of the process in step S4 described above is as follows.
[0057]
a threshold TH is calculated from r (n),
If | r (n) |> TH, then v (n) = k1
If | r (n) | ≦ TH, then v (n) = k2.
Here, k1 and k2 are k1>k2> 0.
In this relationship, a large position weight k1 is set at a position where the absolute value amplitude is large. If k1 = k2, the position weight is not used. Further, although one threshold value TH is used, a method of setting the position weight value more finely using a plurality of types of threshold values by using TH1 and TH2 is also effective.
[0058]
This is the end of the description of the flowchart of FIG.
[0059]
Since the present invention relates to weighting used for parameter selection on the encoding side, the decoding method using the code of each parameter obtained by encoding is the same as the conventional method. Here, the decoding method will be briefly described with reference to FIG.
[0060]
In FIG. 5, the encoded data from the encoding unit is input from the input terminal 160 and is separated into the respective codes A, L, C, and G by the encoded data separation unit 19. The spectrum parameter decoding unit 14 reproduces the spectrum parameter based on the code A. The adaptive excitation decoding unit 11 reproduces an adaptive code vector based on the code L. The noise source decoding unit 12 reproduces a noise code vector based on the code C. The gain decoding unit 13 reproduces the gain based on the code G. The sound source reproduction unit 15 reproduces a sound source signal using the reproduced adaptive code vector, noise code vector, and gain.
[0061]
The synthesis filter 16 constitutes a synthesis filter using the spectrum parameters reproduced by the spectrum parameter decoding unit 14 and passes a sound source signal from the sound source reproduction unit 15 to generate a synthesized speech signal. The post filter 17 performs a post filtering process for shaping the coding distortion included in the synthesized speech signal so that the sound becomes easy to hear. The processed synthesized speech signal is output from the output terminal 195.
[0062]
【The invention's effect】
According to the first aspect of the present invention, it is possible to perform code selection such that encoding distortion at an important position is reduced by introducing position weights into parameter code selection.
[0063]
According to the second aspect of the present invention, it is possible to adaptively set the position weight based on the signal obtained from the input signal.
[0064]
According to the third aspect of the present invention, since position weights are used only at specific positions, it is not necessary to set position weights at all positions, thereby reducing the amount of calculation.
[0065]
According to the fourth aspect of the present invention, in addition to the effect of any one of the first to third aspects, a sound source signal such as a pitch pulse is used in encoding a CELP-type sound source signal by introducing position weighting. Overcoming the problem that the characteristics of can not be expressed well.
[0066]
Further, according to the invention described in claim 5, the effect of the invention described in any one of claims 2 to 4 is obtained.
[0067]
According to the invention of claim 6, in addition to the effect of any one of claims 1 to 5, distortion is more effectively achieved by using both position weights and conventional auditory weights. It is possible to select a code that is difficult to hear.
[0068]
Moreover, according to the invention of Claim 7, the effect of any one of Claims 1 to 6 can be obtained.
[Brief description of the drawings]
FIG. 1 is a block diagram showing a basic configuration of a speech / acoustic signal encoding method according to the present invention.
FIG. 2 is a diagram (No. 1) for explaining a procedure for obtaining a position weight from an audio signal by the method of the present embodiment;
FIG. 3 is a diagram (No. 2) for explaining a procedure for obtaining a position weight from an audio signal by the method of the present embodiment;
FIG. 4 is a flowchart for explaining an encoding method according to an embodiment of the present invention.
FIG. 5 is a diagram for explaining a decoding method;
[Explanation of symbols]
100 input terminals
120 Residual signal calculator
150 encoded data
910 Position weight control unit
911 Spectral parameter codebook search unit
912 Adaptive codebook search unit
913 Noise codebook search unit
914 Gain codebook search unit
915 Encoded data output unit

Claims (7)

An audio / acoustic signal encoding method for generating a synthesized sound signal by passing a sound source signal through a synthesis filter,
A time series signal corresponding to a component that could not be predicted with respect to the input signal is obtained, and at the sample position where the power of the time series signal is large , the distortion between the synthesized sound signal and the target sound signal is further reduced. A weight information acquisition step for acquiring position weight information;
An evaluation step in which the distortion of the synthesized sound signal and the target sound signal is evaluated by weighting using the weight information acquired in the weight information acquisition step;
And a code selection step of selecting a code of a parameter of the sound source signal so that distortion between the synthesized sound signal and the target sound signal is reduced based on a result of the evaluation in the evaluation step. A method for encoding an acoustic signal. An audio / acoustic signal encoding method for generating a synthesized sound signal by passing a sound source signal through a synthesis filter,
A first evaluation step for obtaining a time-series signal corresponding to a component that could not be predicted for the input signal , and evaluating a function value representative of the amplitude or the magnitude of the amplitude of the time-series signal;
A weight information acquisition step of acquiring position weight information for reducing distortion at a sample position having a large amplitude or function value in the first evaluation step;
A second evaluation step of evaluating distortion of the synthesized sound signal and the target sound signal by weighting using the position weight information acquired in the weight information acquisition step;
A code selection step of selecting a code of a parameter of the sound source signal so that distortion between the synthesized sound signal and the target sound signal is reduced based on a result of the evaluation in the second evaluation step. Encoding method of voice / acoustic signal. An audio / acoustic signal encoding method for generating a synthesized sound signal by passing a sound source signal through a synthesis filter,
A time series signal corresponding to a component that could not be predicted with respect to the input signal is obtained, a function value representative of the amplitude or the magnitude of the amplitude of the time series signal is evaluated, and the sample position with the large amplitude or function value First position weight information corresponding to a large weight value is acquired, and second position weight information corresponding to a small weight value, which is different from the first position weight information, with respect to a sample position having a small amplitude or function value A weight information acquisition step for acquiring
An evaluation step of weighting and evaluating distortion of the synthesized sound signal and the target sound signal using the first and second weight information acquired in the weight information acquisition step;
And a code selection step of selecting a code of a parameter of the sound source signal so that distortion between the synthesized sound signal and the target sound signal is reduced based on a result of the evaluation in the evaluation step. A method for encoding an acoustic signal.   The voice according to any one of claims 1 to 3, wherein the parameter of the sound source signal is information for searching an adaptive codebook or a noise codebook used for obtaining the sound source signal. / Encoding method of acoustic signal. Time-series signal obtained from the input signal, the prediction residual signal, a voice / sound according to claim 1, one of the 4, characterized in that any of the simulated signal of the predictive residual signal Signal encoding method.   6. The speech / acoustic signal encoding method according to claim 1, wherein a parameter of the sound source signal is selected using the position weight information and auditory weight. An input unit for inputting voice / acoustic signals;
An encoding unit that performs an encoding process on a voice / acoustic signal input via the input unit;
A transmission unit for transmitting the voice / acoustic signal encoded by the encoding unit;
A receiver for receiving the encoded voice / acoustic signal;
A decoding unit that performs a decoding process on the voice / acoustic signal received through the receiving unit;
An output unit for outputting the voice / acoustic signal decoded by the decoding unit;
An electronic apparatus, wherein the encoding unit executes the encoding method according to any one of claims 1 to 6.

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