JP3299099B2 - Audio coding device - Google Patents

Abstract

PROBLEM TO BE SOLVED: To provide a voice encoding device with less amount of search operation and memory size and less degradation of sound quality by providing a function to search at least one of code vectors while shifting its position, when a sound source quantizing part searches for a code book. SOLUTION: This device is provided with a spectral parameter calculation part 4 to obtain a spectral parameter from an input voice signal to quantize, and a sound source quantizing part 12 to search fox a code book 13 storing a sound source signal of the voice signal beforehand and quantizes to output it. And when a sound source quantizing part 12 searches for a sound source code book 13, it searches for it while shifting a sample position of at least one of the code vectors. And if a size of the entire code book is expressed by B bits and a shift amount is expressed by A bits, the size of the code book to be stored becomes not B bits but B-A bits and this can decreases a memory size necessary for a storage.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a speech coding apparatus, and more particularly to a speech coding apparatus for coding a speech signal with a relatively small amount of computation and a small amount of memory in high quality.

[0002]

2. Description of the Related Art A speech encoding device is used opposite to a speech decoding device, and a speech decoding device decodes speech encoded by the speech encoding device. Here, as a method of encoding a speech signal with high efficiency, for example, M. Schroeder and B. Atal
IEEE Proceedings (IEEE Pro
c.) ICASSP-85, pp. 937-940, Code Excited Linear Prediction: High Quality Speech at Very Low Bit Rates.
Peech at very lowbit rates) (1), and Claizin (Kleijn) et al., IEEE Proc. ICASSP-88,
Improve Speech Quality and Efficient Vector Quantization in SLP, 1988, pp. 155-158.
d speech quality and efficient vector quantization
in CELP (Code Excited Linear Prediction Co.)
ding) is known. In this method, the transmitting side performs linear prediction (LP) from a speech signal every frame (for example, 20 ms).
C) Using analysis, extract a spectral parameter representing a spectral characteristic of the audio signal, further divide the frame into a plurality of subframes (for example, 5 ms), and apply an adaptive codebook to each subframe based on a past sound source signal. (Delay parameter and gain parameter corresponding to the pitch period) are extracted, the speech signal of the corresponding subframe is pitch-predicted by the adaptive codebook, and the residual signal obtained by pitch prediction is subjected to sound source quantization. The unit stores an excitation codebook (vector quantization codebook) composed of a predetermined type of noise signal, selects an optimal excitation codevector from this codebook, and calculates an optimal gain. Quantize the sound source signal.
The excitation code vector is selected so as to minimize the error power between the signal synthesized from the selected noise signal and the above-mentioned residual signal. Then, an index and a gain indicating the type of the selected code vector, a spectrum parameter and a parameter of the adaptive codebook are combined and transmitted by the multiplexer unit. Description on the receiving side is omitted.

[0003]

The above-mentioned conventional speech coding apparatus requires a bit rate of 8 kb / s or more in order to obtain good sound quality. This means that the number of bits in the sound source codebook is, for example, 10 per 5 ms subframe.
A bit more codebook was needed.
For this reason, there is a problem that a large amount of calculation and a large amount of memory are required for searching and storing the sound source codebook. For example, considering a 10-bit codebook with a 5 ms subframe, the search with the simplest square distance requires 1024x40x200 = 8,192,000 multiplications per second, and the memory amount is 1024x40 = 40,240 words. Needed. On the other hand, when the number of bits is reduced in order to reduce the amount of calculation and the amount of memory, there is a problem that sound quality is deteriorated.

In the above-mentioned conventional speech coding apparatus, a codebook having a large number of bits is required in order to obtain good coded sound quality. The point is that it can take a great phase. Therefore, in order to express these different phases as a pattern of a sound source code vector, a codebook of a certain scale or more is required.

SUMMARY OF THE INVENTION It is an object of the present invention to solve the above-mentioned problems and to provide a speech coding apparatus with less deterioration in sound quality with a smaller amount of search operation and less memory than conventional methods.

[0006]

According to the present invention, there is provided a speech coding apparatus comprising: a spectrum parameter calculating section for obtaining a spectrum parameter from an input speech signal and quantizing the spectrum parameter; and using the spectrum parameter to generate a sound source signal of the speech signal. In a speech encoding apparatus having a sound source quantization unit for searching, quantizing and outputting a codebook stored in advance, the sound source quantization unit stores the codebook when searching for the codebook. This is a configuration having a function of searching for at least one of the code vectors while shifting the position.

A speech coding apparatus according to the present invention comprises a spectrum parameter calculation unit for obtaining and quantizing a spectrum parameter from an input speech signal, and a codebook in which a sound source signal of the speech signal is stored in advance using the spectrum parameter. A sound source quantization unit that searches for, quantizes, and outputs an audio signal. A mode discrimination unit that discriminates a mode from an audibility weighted signal obtained by performing audibility weighting on the audio signal in frame units and outputs mode information. When,
In a predetermined mode when the sound source quantization unit searches the codebook, the sound source quantization unit has a function of searching for at least one of the code vectors stored in the codebook while shifting the position.

A speech coding apparatus according to the present invention comprises a spectrum parameter calculating section for obtaining and quantizing a spectrum parameter from an input speech signal, and a codebook in which a sound source signal of the speech signal is stored in advance using the spectrum parameter. A sound source quantization unit that searches for, quantizes, and outputs an audio signal. A mode discrimination unit that discriminates a mode from an audibility weighted signal obtained by performing audibility weighting on the audio signal in frame units and outputs mode information. When,
When the sound source quantization unit searches the codebook, the sound source quantization unit has a function of performing a search while changing an amount to shift a position of at least one of the code vectors stored in the codebook according to the mode information. is there.

A speech coding apparatus according to the present invention comprises a spectrum parameter calculation unit for obtaining and quantizing a spectrum parameter from an input speech signal, and a code which previously stores a sound source signal of the speech signal using the spectrum parameter. A sound source quantization unit for searching, quantizing, and outputting a book, wherein a value determined for each code vector stored in the code book when the excitation quantization unit searches the code book. And a function of searching while changing the amount by which the position is shifted in accordance with

[Operation] In the first invention, when the excitation codebook is searched in the excitation quantization unit, the search is performed while shifting the sample position of at least one code vector. For the sake of simplicity, it is assumed that the search is performed while shifting all the code vectors. If the entire codebook size is represented by B bits and the shift amount is represented by A bits, the codebook size to be stored is B bits. Instead, the number of bits becomes BA, and the memory required for storage can be reduced. Therefore, it is clear that even when shifting some code vectors, the memory amount can be reduced as compared with the conventional method.

Next, a method of searching for a sound source codebook will be described. For the search, for example, an autocorrelation approximation method is used. In this method, a sound source code vector c _k (n) that maximizes the second term on the right side of the following equation is selected.

[0012]

Where

[0014]

## EQU1 ## further

[0016]

## EQU1 ## Details of this method are described in I. Trancoso et al., IEEE Proc. ICASSP-86, 1986, 2375
Efficient Procedure for Finding the Uptime Innovation in Statistical Coders on page 2378
ures for finding the optimum innovation in stochas
The description is omitted because a paper (Reference 3) entitled "tic coders" can be referred to.

Here, for the phase-shifted code vector component, no calculation is necessary because the denominator value _Pk is the same. Therefore, the amount of calculation required for calculating the denominator is reduced by the number A of bits of the shift.

In the second invention, a feature amount is obtained from input speech in a predetermined time section (hereinafter referred to as a frame), and the speech of the frame is classified into one of a plurality of types of modes. In the following, there are four modes,
This is represented by two bits as mode information and transmitted. In the case of the predetermined mode, the excitation quantization section searches while shifting the sample position of at least one code vector when searching the excitation codebook. The method of searching while shifting the code vector is the same as in the first invention.

According to a third aspect, in the second aspect, the shift amount A of the sample position is changed for each mode. For example, A is 0 bits in mode 0,
The value takes 5 bits in mode 1, 4 bits in mode 2, and 3 bits in mode 3.

According to a fourth aspect of the present invention, in the excitation quantization section of the first invention, when searching the excitation codebook, the search is performed while changing the shift amount of the sample position according to the code vector. However, the total shift amount of the entire codebook is a fixed value, for example, A bits.

[0022]

Next, embodiments of the present invention will be described with reference to the drawings.

FIG. 1 is a block diagram showing a first embodiment of the present invention.

A speech encoding apparatus 1 according to a first embodiment of the present invention comprises a frame dividing circuit 2 for dividing an inputted speech signal into frames of a predetermined time length, and a speech signal of a frame having a shorter time than the frame. A sub-frame division circuit 3 for dividing into a long sub-frame, and receiving at least one audio signal of a series of frames output from the frame division circuit 2
A spectrum parameter calculation circuit 4 that cuts out the audio signal by applying a window longer than the subframe time length to the audio signal of one subframe and calculates spectral parameters to a predetermined order, and a line spectrum versus parameter codebook (Hereinafter referred to as LSP codebook) 6
A parameter quantization circuit 5 for vector-quantizing an LSP parameter quantized in a predetermined subframe calculated by the spectrum parameter calculation circuit 4 using
A perceptual weighting circuit 7 that receives the linear prediction coefficients of a plurality of subframes calculated by the spectrum parameter calculation circuit 4 and weights the perceptual weight of the audio signal of each subframe and outputs a perceptual weighting signal; Are input for each sub-frame, the linear prediction coefficients restored by the spectral parameter quantization circuit 5 are calculated for each sub-frame, the response signal is calculated for one sub-frame, and output to the subtractor 8 A signal calculation circuit 9, an impulse response calculation circuit 10 that receives the linear prediction coefficient restored by the spectrum parameter quantization circuit 5 and calculates a predetermined score of an impulse response of the perceptual weighting filter, and a past sound source signal that returns from the output side. The output signal of the subtractor 8 and the impulse response of the perceptual weighting filter are input and An adaptive codebook circuit 11 for outputting a Intex representing a delay determined delay corresponding to the switch,
A sound source quantization circuit 12 that quantizes a sound source signal using a sound source code book 13 and a gain code vector are read from a gain code book 15 to select an optimal gain code vector, and an index representing the selected gain code vector is set as an index. It comprises a gain quantization circuit 14 that outputs to the multiplexer 16, and a weighting signal calculation circuit 17 that receives the output of the gain quantization circuit 14 and reads a code vector corresponding to the output from the index to obtain a drive excitation signal.

Next, the operation of the present apparatus will be described.

First, an audio signal is input from an input terminal, and the frame dividing circuit 2 converts the audio signal into a frame (for example, 10 m).
s), and the subframe division circuit 3 divides the audio signal of the frame into subframes (for example, 2.5 ms) shorter than the frame. In the spectrum parameter calculation circuit 4, a window (for example, 24 ms) longer than the subframe length is applied to the audio signal of at least one subframe.
, The speech is cut out, and the spectral parameters are calculated in a predetermined order (for example, P = 10th order). Here, a well-known LPC analysis, a Burg analysis, or the like can be used for calculating the spectrum parameters. Here, Burg analysis is used. Berg (B
urg) For details of the analysis, see the book “Corona Analysis and System Identification” written by Nakamizo (Corona Publishing Co., 1988).
The description is omitted since it is described on pages 82 to 87 (Document 4) and the like.

Further, in the spectrum parameter calculation circuit 4, the linear prediction coefficient α calculated by the Burg method
_i (i = 1,..., 10) is converted to LSP parameters suitable for quantization and interpolation. Here, the conversion from the linear prediction coefficient to the LSP is performed by Sugamura et al. In a paper entitled "Speech Information Compression by Line Spectrum Pair (LSP) Speech Analysis / Synthesis Method" (Transactions of the Institute of Electronics, Information and Communication Engineers, J64-A, pp.599). -606, 1981) (Reference 5)
Can be referred to. For example, the linear prediction coefficient obtained by the Burg method in the second and fourth subframes is represented by LS
Convert to P parameter, LSP of 1st and 3rd subframe
Is obtained by linear interpolation, and the LS of the first and third sub-frames is calculated.
P is inversely transformed back to the linear prediction coefficient, and the linear prediction coefficients α _il i = 1,..., 10, l = 1,. Also, the LS of the fourth sub-frame
P is output to the spectrum parameter quantization circuit 5.

In the speckle parameter quantization circuit 5, L
The LSP parameter of a predetermined subframe is efficiently vector-quantized using the SP record book 6, and a quantized value that minimizes the distortion of the following equation is output.

[0029]

Here, LSP (i), QLSP (i) _j and W (i) are the i-th LSP and LSP codebook 6 before quantization, respectively.
Are the j-th code vector and the weight coefficient.

In the following, it is assumed that the LSP parameter of the fourth sub-frame is quantized. A well-known method can be used for the method of vector quantization of LSP parameters. Specific methods are described in, for example, JP-A-4-171500 (Reference 6), JP-A-4-363000 (Reference 7), and JP-A-5-6199 (Reference 8).
And IEE Proceedings by T. Nomura and others. Mobile multimedia
Communications (IEEE Proc. Mobile Multimedia
Communications.) 1993; On page 2.5, LSP Coding Using Viku-SV-Q-with with Interpolation in 4.0
75KPS M-LSP Coding Using VQ-SVQ Wit
h Interpolation in 4.075 kbps M-LCELP Speech Code
r), the description of which is omitted here.

The spectrum parameter quantization circuit 5
Then, the LSP parameters of the first to fourth subframes are restored based on the LSP parameters quantized in the fourth subframe. Here, the quantization LSP parameter of the fourth sub-frame of the current frame and the fourth
The LSPs of the first to third subframes are restored by linearly interpolating the quantized LSPs of the subframe. Here, after selecting one type of code vector that minimizes the error power between the LSP before quantization and the LSP after quantization, the LSPs of the first to fourth subframes can be restored by linear interpolation. In order to further improve performance, after selecting a plurality of candidates for the code vector that minimizes the error power, evaluate the cumulative distortion for each candidate, and select a combination of the candidate and the interpolation LSP that minimizes the cumulative distortion. You can make it. For details, see, for example, Japanese Patent Application No. 5-8737 (Reference 10).
Can be referred to.

The LSPs of the first to third sub-frames and the quantized LSPs of the fourth sub-frame, which have been reconstructed as described above, are converted into linear prediction coefficients α ′ _il (i = 1,..., 10, l =,. ) And outputs it to the impulse response calculation circuit 10. Further, an index representing the code vector of the quantized LSP of the fourth subframe is output to the multiplexer 16. The audibility weighting circuit 7 includes a spectrum parameter calculation circuit 4
From the linear prediction coefficient α before quantization for each subframe.
_il (i = 1,..., 10, l =,..., 5) is input, and the perceptual weighting is performed on the audio signal of the subframe based on Document 1 to output a perceptual weighting signal.

The response signal calculation circuit 9 receives the linear prediction coefficient α _il for each subframe from the spectrum parameter calculation circuit 4, and quantizes, interpolates and restores the linear prediction coefficient α _il from the spectrum parameter quantization circuit 5. α ′ _il is input for each sub-frame, a response signal with the input signal set to zero d (n) = 0 is calculated for one sub-frame using the stored value of the filter memory, and output to the subtractor 8 I do. Here, the response signal x _z (n) is represented by the following equation.

[0035]

However, when ni ≦ 0, y (ni) = p (N + (ni)) (8) x _z (ni) = s _w (N + (ni)) (9) where N is the subframe length Is shown. γ is a weight coefficient for controlling the perceptual weighting amount, and is the same value as the following equation (11). s _w (n) and p (n) represent the output signal of the weighting signal calculation circuit 17 and the output signal of the denominator term of the first term filter on the right-hand side in Expression (11) described later, respectively.

The subtractor 8 subtracts the response signal by one subframe from the perceptual weighting signal according to the following equation, and outputs x ′ _w (n) to the adaptive codebook circuit 11. x ′ _w (n) = x _w (n) −x _z (n) (10) The impulse response calculation circuit 10 calculates in advance the impulse response h _w (n) of the auditory weighting filter whose z-transformation is expressed by the following equation. Calculation is performed for a predetermined number L, and the adaptive codebook circuit 11, the sound source quantization circuit 12, and the gain quantization circuit 14
And output to

[0038]

In the adaptive code book circuit 11, the past sound source signal v (n) is output from the gain quantization circuit 14, the output signal x ′ _w (n) is output from the subtractor 8, and the audibility is output from the impulse response calculation circuit 10. Enter the weighted impulse response h _w (n). The delay T corresponding to the pitch is determined so as to minimize the distortion of the following expression, and an index representing the delay is output to the multiplexer 16.

[0040]

Here, y _w (n−T) = v (n−T) * h _w (n) (13), and the symbol * represents a convolution operation. The gain β is obtained according to the following equation.

[0042]

Here, in order to improve the accuracy of extracting delays for female sounds and children's voices, the delays may be determined by decimal sample values instead of integer samples. A concrete method is, for example, IACSSP by P. Kroon et al. (IEEE Proc.)
-90, 1990, Pitch Predictors with High Temporal Solution (Pit
For example, a paper (Reference 11) published under the title of "Ch predictors with high temporal resolution" can be referred to.

Further, the adaptive codebook circuit 11 performs pitch prediction according to the following equation, and outputs a prediction residual signal e _w (n) to the sound source quantization circuit 12. e _w (n) = x ′ _w (n) −βv (nT) * h _w (n) (15) The sound source quantization circuit 12 has a feature in the search of the sound source codebook as described in the operation.

FIG. 2 is a block diagram showing the configuration of the sound source quantization circuit in FIG.

In the following description, it is assumed that the index to be transmitted of the entire sound source codebook is B bits and the shift amount is A bits.

The inverse filtering circuit 18 of the sound source quantization circuit 12 receives the adaptive codebook prediction residual signal e _w (n) and the perceptual weighting impulse response h _w (n) and calculates the following equation.

[0048]

The sound source code book 13 has a size of (BA) bits. The autocorrelation calculation circuit 20 reads out the sound source code vector c _k (n) from the sound source codebook 13 and calculates the autocorrelation using the following equation. This value is commonly used for a code vector whose position is shifted with respect to the same code vector.

[0050]

Also, the autocorrelation of the auditory weighting impulse response is calculated by the following equation.

[0052]

The position shift circuit 19 sequentially shifts the position of the excitation code vector c _k (n) by the following equation. c _kl (n) = c _k (n + 1), l = 0,..., 2 ^A −1 (19) Here, A represents the number of bits for representing the shift amount. The cross-correlation calculation circuit 21 calculates a cross-correlation according to the following equation.

[0054]

The square calculation circuit 22 calculates the square of the cross-correlation CC. The division circuit 23 outputs the result of the division between CC ² and P _{k in} Equation (2) to the maximum value discrimination circuit 24. The maximum value discriminating circuit 24 discriminates the maximum of the division result, and outputs to the gain quantization circuit 14 a total index in which the index of the sound source codebook 13 and the shift amount are added at that time.

The gain quantization circuit 14 reads a gain code vector from the gain code book 15 and selects a gain code vector for the selected excitation code vector so as to minimize the following equation. Here, an example in which both the gain of the adaptive codebook and the gain of the sound source are simultaneously vector-quantized will be described.

[0057]

Here, β ′ _k and G ′ _k are k-th code vectors in the two-dimensional gain codebook stored in the gain codebook 15. An index representing the selected gain code vector is output to the multiplexer 16.

The weighting signal calculation circuit 17 receives the output parameters of the spectrum parameter calculation circuit 4 and the respective indexes, reads out the corresponding code vectors from the indexes, and firstly obtains the driving sound source signal v (n) based on the following equation. Ask for. v (n) = β ′ _k v (nT) + G ′ _k c _k (n) (22) v (n) is output to the adaptive codebook circuit 11.

Next, using the output parameter of the spectrum parameter calculation circuit 4 and the output parameter of the spectrum parameter quantization circuit 5, the response signal (s
_w (n)) is calculated for each subframe, and output to the response signal calculation circuit 9.

[0061]

Thus, the description of the first embodiment of the present invention is completed.

FIG. 3 is a block diagram showing a second embodiment of the present invention.

Speech Encoding Apparatus 2 of Second Embodiment
5 is different from the first embodiment in that a mode discrimination circuit 26 is newly provided and a part of the function of a sound source quantization circuit 27 is changed. The other components having the same reference numerals as those in FIG. 1 perform the same operations as those in FIG.

The mode discriminating circuit 26 receives the perceptual weighting signal from the perceptual weighting circuit 17 in frame units, and outputs mode discriminating information. Here, the feature amount of the current frame is used for mode determination. As the characteristic amount, for example, a pitch prediction gain averaged in a frame is used. For example, the following equation is used for calculating the pitch prediction gain.

[0066]

Here, L is the number of subframes included in the frame. P _i and E _i indicate the speech power and the pitch prediction error power in the i-th subframe, respectively.

[0068]

Here, T is an optimal delay for maximizing the prediction gain.

Next, the frame average pitch prediction gain G
Is compared with a plurality of predetermined threshold values, and classified into a plurality of types of modes. As the number of modes, for example, 4 can be used. The mode determination circuit 26
The mode determination information is output to the sound source quantization circuit 27 and the multiplexer 16.

When the mode discrimination information indicates a predetermined mode, the sound source quantization circuit 27 searches while shifting the sound source code vector.

FIG. 4 is a block diagram showing the configuration of the sound source quantization circuit in FIG. The sound source quantization circuit 27 differs from the sound source quantization circuit 12 in that a part of the function of the position shift circuit 28 is changed. The other components having the same reference numerals as those in FIG. 2 perform the same operations as those in FIG.

The position shift circuit 28 includes a mode discriminating circuit 2
The mode information is input from step 6 to shift the position of the excitation code vector in the case of a predetermined mode. Subsequent operations are the same as those of the position shift circuit 19 in FIG. This concludes the description of the second invention.

FIG. 5 is a block diagram showing a third embodiment of the present invention.

Speech coding apparatus 2 according to a third embodiment
9 differs from the second embodiment in that a part of the function of the sound source quantization circuit 30 is changed. The other components having the same reference numerals as those in FIG. 3 perform the same operations as those in FIG.

FIG. 6 is a block diagram showing the configuration of the sound source quantization circuit in FIG. The sound source quantization circuit 30 differs from the sound source quantization circuit 27 in that a part of the function of the position shift circuit 31 is changed. The other components denoted by the same reference numerals as those in FIGS. 2 and 4 perform the same operations as those in FIGS.

The position shift circuit 31 inputs the mode information as described in the section of operation, and changes the shift amount of the position of the code vector for each mode. That is, the number of bits A required for the shift is changed for each mode. For example,
Mode 0, 0 bits, A ₁ bit in mode 1, A ₂ bit in mode 2, the value A _3-bit in mode 3 take. FIG. 7 is a block diagram showing a fourth embodiment of the present invention.

Speech coding apparatus 3 according to a fourth embodiment
2 is different from the first embodiment in that a part of the function of the sound source quantization circuit 33 is changed. The other components having the same reference numerals as those in FIG. 1 perform the same operations as those in FIG.

FIG. 8 is a block diagram showing the configuration of the sound source quantization circuit in FIG. The difference between the sound source quantization circuit 12 and the sound source quantization circuit 33 is that the assignment circuit 34 assigns an amount to shift the position according to the index of the code vector stored in the sound source codebook 13. However, the total shift amount is determined as A bits in the entire codebook. When the shift amount is input from the assignment circuit 34, the position shift circuit 35 shifts the position of the code vector.

The description of the embodiment of the present invention has been completed.

The present invention is not limited to the above-described embodiment, but can be variously modified. For example, the sound source codebook may have a configuration as conventionally used, or may have a configuration including a plurality of pulse trains. The code vector of the sound source codebook may be configured by learning in advance using audio signal data. Furthermore, it is also possible to adopt a configuration in which the adaptive codebook circuit, the sound source codebook, and the gain codebook are switched using the mode determination information.

[0082]

As described above, according to the present invention, when searching for a codebook, the sound source quantization unit searches for at least one of the code vectors stored in the codebook while shifting the position. By and
By changing the shift amount for each mode,
Even with the same bit rate as the conventional method, there is an effect that both the amount of calculation required for searching the codebook and the amount of memory required for storing the codebook can be reduced. Also,
This effect is also increased by increasing the number of bits spent for shifting.

[Brief description of the drawings]

FIG. 1 is a block diagram showing a first embodiment of the present invention.

FIG. 2 is a block diagram showing a configuration of a sound source quantization circuit in FIG.

FIG. 3 is a block diagram showing a second embodiment of the present invention.

FIG. 4 is a block diagram illustrating a configuration of a sound source quantization circuit in FIG. 3;

FIG. 5 is a block diagram showing a third embodiment of the present invention.

FIG. 6 is a block diagram showing a configuration of a sound source quantization circuit in FIG. 5;

FIG. 7 is a block diagram showing a fourth embodiment of the present invention.

FIG. 8 is a block diagram showing a configuration of a sound source quantization circuit in FIG. 7;

[Explanation of symbols]

1, 25, 29, 32 Speech coding apparatus 2 Frame division circuit 3 Subframe division circuit 4 Spectrum parameter calculation circuit 5 Spectrum parameter quantization circuit 6 Line spectrum pair parameter codebook (LSP
Codebook) 7 auditory weighting circuit 8 subtractor 9 response signal calculation circuit 10 impulse response calculation circuit 11 adaptive codebook circuit 12, 27, 30, 33 sound source quantization circuit 13 sound source codebook 14 gain quantization circuit 15 gain codebook 16 Multiplexer 17 Weighting signal calculation circuit 18 Inverse filtering circuit 19, 28, 31, 35 Position shift circuit 20 Autocorrelation calculation circuit 21 Cross-correlation calculation circuit 22 Square calculation circuit 23 Division circuit 24 Maximum value determination circuit 26 Mode determination circuit 34 Assignment circuit

 ────────────────────────────────────────────────── ─── Continuation of the front page (56) References JP-A-2-79100 (JP, A) JP-A-5-281999 (JP, A) JP-A-6-222797 (JP, A) JP-A-6-222797 266395 (JP, A) Special table Hei 2-501166 (JP, A)

Claims (4)

(57) [Claims] A spectrum parameter calculator for calculating and quantizing a spectrum parameter from an input speech signal;
A sound source quantization unit that searches for, quantizes, and outputs a codebook in which a sound source signal of the sound signal is stored in advance using the spectrum parameter, wherein the sound source quantization unit includes the codebook. When searching for, a part of the code vectors stored in this codebook is searched while shifting the position , and the
Search for other code vectors without shifting the position.
A speech encoding device having a search function. 2. A spectrum parameter calculation unit for obtaining and quantizing a spectrum parameter from an input speech signal,
A sound source quantization unit for searching, quantizing, and outputting a code book in which a sound source signal of the sound signal is stored in advance by using the spectrum parameter; and A mode discriminator that discriminates a mode from the perceived weighting signal and outputs mode information, and a code vector stored in the code book in a predetermined mode when the sound source quantization unit searches the code book. We searched while shifting the <br/> position for some in the other code vectors
An audio coding apparatus having a function of searching for a file without shifting its position . 3. A spectrum parameter calculator for obtaining and quantizing a spectrum parameter from an input speech signal,
A sound source quantization unit that searches for a code book in which the sound source signal of the sound signal is stored in advance by using the spectrum parameter, and quantizes and outputs the code book. A mode discrimination unit that discriminates a mode from the perceived weighting signal that has been subjected to and outputs mode information, and a position of at least one of the code vectors stored in the codebook when the sound source quantization unit searches the codebook. A function of performing a search while changing an amount by which to shift according to the mode information. 4. A spectrum parameter calculator for obtaining and quantizing a spectrum parameter from an input speech signal;
A sound source quantization unit that searches, quantizes, and outputs a codebook in which a sound source signal of the sound signal is stored in advance using the spectrum parameter, wherein the sound source quantization unit includes the codebook. A speech encoding apparatus having a function of performing a search while changing an amount of position shift according to a value determined for each code vector stored in the codebook when searching for.