WO1999038157A1

WO1999038157A1 - Method and device for coding lag parameter and code book preparing method

Info

Publication number: WO1999038157A1
Application number: PCT/JP1999/000294
Authority: WO
Inventors: Koji Yoshida
Original assignee: Matsushita Electric Industrial Co., Ltd.
Priority date: 1998-01-27
Filing date: 1999-01-26
Publication date: 1999-07-29
Also published as: CN1256001A; KR20010005669A; CA2283203A1; EP0971338A1; AU2075199A

Abstract

A lag parameter coding means (215b) generates a code corresponding to a lag parameter value by using a lag parameter code book (215a). On the decoding side, a lag parameter value corresponding to the lag parameter code generated on the coding side is decoded by using the same lag parameter code book (215a) and outputted. In the lag parameter code book (215a), the relation between the lag parameter value and the corresponding code (P) is shown. The relation is so determined as to increase the rate at which the decoded lag parameter value of when a bit error occurs in the code deviates to approximately an integral multiple (including one time) or an integral submultiple of the decoded lag parameter value of when no bit error occurs. As a result, the auditory degradation of quality of decoded sound is suppressed even when the code has a bit error.

Description

TECHNICAL FIELD The present invention relates to a lag parameter encoding method and apparatus, and a codebook creating method.

The present invention relates to a sound processing device applicable to a digital cellular phone, a personal computer, and the like, and more particularly, to a lag parameter including a pitch period, which is one of parameters representing characteristics of a sound signal, or a parameter related thereto. The present invention relates to a method and an apparatus for encoding a lag parameter to be encoded, and a method for creating a codebook used for these. Background art

Important parameters that express the characteristics of the audio signal include the pitch period and lag parameter of the audio signal. These parameters are used as coding parameters in voice coding processing for coding voice signals with high efficiency, and as synthesis parameters in voice synthesis. When transmitting or storing a lag parameter, the parameter value must be encoded into a code corresponding to the value according to a specific rule. The coding method of the lag parameter in voice coding is described in Recommendation G.729 (8 kbps CS-ACELP voice coding method) of International Organization ITU-T.

The lag parameters coded according to the recommendation are transmitted with the codes of the other coding parameters. The lag parameter in this conventional example has a codebook called an adaptive codebook when generating the excitation signal used for the synthesis of decoded speech in the CS-ACELP system, which is the speech coding system of this conventional example. It is a value (lag value) that indicates which section of the signal to use. This lag value T consists of an integer part T 1 (T 1 = 19 to 143) and a decimal part frac / 3 (frac = _l, 0 , 1).

The lag value T is calculated by the encoding device using the above T l and frac as follows: It is encoded as a code P (P = 0 to 255) by the equation.

(3 X (Tl-19) + frac-1, Tl = 19 to 85, frac =-1,0,1

P =

l (Tl-85) +197, Tl = 86 ~ 143, frac = 0

(1) On the other hand, decoding is performed by the decryption device based on the decoding lag value T 1, frac and code P according to the inverse rule of equation (1).

The lag parameter is a delay amount from a certain time t1 of the audio signal to a time t0 before the time t1 and similar to the waveform at the time t1. That is, the lag parameter is typically a parameter representing a pitch cycle in a periodic waveform, and is the pitch cycle of the voice itself. However, lag parameter is a broad concept that includes the pitch period, in the sense that it also includes the amount of delay to a position where the waveform is similar in a non-periodic voice waveform, such as a rising section of voice. .

However, the lag parameter code obtained by the conventional lag parameter coding method described above, if a bit error occurs during transmission or storage, the decoded lag value is significantly different from the correct lag value without error, and There is a possibility that large deterioration will occur.

In general, as one method of suppressing quality degradation due to bit errors in codes, distortions such as Euclidean distance between parameter values of coding parameters and inter-code distance (hamming distance) between codes representing the parameter values are used. There is a method to reduce the effects of bit errors by giving some correlation to and.

When the Euclidean distance between the lag values and the difference value are used as the distortion measure between the lag parameter values, it is effective in the range where the values are small. However, if the value exceeds a certain value, it will not be possible to cope with audible distortion. Therefore, it is not very effective to adopt the above-mentioned general method for encoding and decoding the lag parameter. As a countermeasure against such a bit error, there is a method of detecting a bit error and not using a lag value containing the error. However, this method is complicated in error detection itself and requires low error such as voice communication. It is not appropriate to add redundant bits such as check bits to the bit rate communication method.

The present invention has been made in view of the above-described circumstances, and has an excellent lag parameter that can suppress deterioration of audio quality due to the error when a bit error occurs in the lag parameter code. It is an object of the present invention to provide an encoding method and apparatus thereof and a codebook creating method. Disclosure of the invention

In order to solve the above-mentioned problem, the present invention encodes a lag parameter using a codebook defined as follows. When a bit error occurs in the lag parameter code, the codebook uses an integer multiple (including 1) or 1 / integer value of the decoding lag parameter value when the decoding lag parameter value has no bit error. It is set to increase the error rate in the vicinity.

The codebook sets the sum of decoded value distortion between codes having a Hamming distance within a specific number of bits to a minimum value or a value close to the minimum, and decodes the decoded lag parameter value as the decoded value distortion between codes. It is generated using a distortion measure such that the distortion is evaluated to be small at an integral multiple or a fraction of an integer.

As a result, the decoding lag parameter value when a bit error occurs in the code is erroneous near an integer multiple (including 1) or a fraction of the decoding lag parameter value when there is no bit error. Since the ratio can be increased, audio signal encoding and decoding can be performed while suppressing deterioration in audio quality in terms of auditory perception. BRIEF DESCRIPTION OF THE FIGURES

FIG. 1 shows a lag parameter encoding method according to Embodiment 1 of the present invention and its encoding method. Schematic block diagram of a wireless transmission device to which the device is applied,

FIG. 2 is a schematic block diagram of a voice coding unit of the wireless communication device according to the first embodiment, FIG. 3 is a main block diagram of a voice coding unit of the wireless communication device according to the first embodiment, and FIG. , A block diagram of a main part of the voice decoding unit of the wireless communication apparatus according to the first embodiment,

FIG. 5 is a diagram showing a procedure of a codebook creating method applied to the wireless communication device according to the first embodiment. BEST MODE FOR CARRYING OUT THE INVENTION

(Embodiment 1)

Hereinafter, Embodiment 1 of the present invention will be described with reference to FIGS.

FIG. 1 is a schematic block diagram of a wireless transmission device to which the present invention is applied. The voice transmission process is performed as follows. The audio signal input from the microphone 101 is A / D converted by the AZD converter 102, output to the audio encoding unit 103, and encoded by, for example, the CELP method. The coded output is modulated by a modulation / demodulation unit 104, for example, according to the CDMA system or the like, and transmitted via a radio transmission unit 105 and an antenna 106.

The voice reception process is performed as follows. The modulated signal received via the antenna 107 and the radio receiving unit 108 is demodulated by the modem 104 and further decoded by the audio decoding unit 109, and is then decoded by the DZA converter 110. After DZA conversion, audio is output from speaker 1 1 1.

The present invention is applied to a part of the adaptive codebook search processing used in the voice coding unit 103 and the voice decoding unit 109 of the wireless communication device.

FIG. 2 is a schematic block diagram of the speech encoding unit 103 of the wireless communication device, and shows a general configuration of a CELP-type speech codec. The A / D-converted audio signal is input from terminal 201 and output to LPC analyzer 202. LPC analysis W section 202 performs linear prediction analysis on the input speech signal, and outputs linear prediction coefficients. The LPC parameter quantization unit 203 quantizes the linear prediction coefficient, and outputs the quantization result to the synthesis filter 204 and the multiplexer 205.

The synthesis filter 204 forms a filter having predetermined characteristics based on the linear prediction coefficients, filters the sound source signal input from the adder 206, and outputs the result to the adder 207. The adder 2007 calculates an error between the input voice signal from the terminal 201 and the output from the synthesis filter 204, and outputs the error signal to the auditory weighting unit 208. The auditory weighting unit 208 performs a weighting process corresponding to the auditory sense on the error signal, and outputs the error signal to the error minimizing unit 209.

The error minimizing unit 209 sets the vector of the adaptive codebook 210 and the fixed excitation codebook 211 so that the error signal output from the auditory weighting unit 208 is minimized, and Gain codebook 2 1 2 Set the gain.

The adaptive codebook 210 is a sound source signal table that accumulates past sound source vectors in predetermined frame units, and includes a plurality of codes according to the vector set by the error minimizing unit 209. And selectively output a specific code string. Multiplier 2 13 multiplies its output by the gain set by gain codebook 2 12, and outputs the result to adder 206.

The adaptive codebook 210 is composed of a buffer for holding the history of the excitation vector output of the adder 206 finally determined for a certain period, and is set by the error minimizing unit 209. A lag value indicating which section of the signal sequence stored in the buffer is to be cut out according to the vector value is output to the lag parameter encoding unit 215. The lag parameter coding unit 2 15 is composed of a lag parameter codebook 2 15 a and a lag parameter coding unit 2 15 b created in advance according to a predetermined rule. The value output is encoded under a certain rule and output to the multiplexer 205. The lag parameter encoding unit 215 will be described later in detail.

In addition, fixed excitation codebook 2 1 1 is the vector set by error minimizing section 2 09. A specific fixed excitation code sequence is selectively output from a plurality of codes according to the torque value. Multiplier 2 16 multiplies its output by the gain set by gain codebook 2 12, and outputs the result to adder 206.

The adder 206 adds the outputs of the multipliers 2 13 and 2 16 and outputs the result to the synthesis filter 204 as a sound source vector. At the same time, the output is fed back to the adaptive codebook 210 and accumulated sequentially.

In this way, error minimizing section 209 measures error signals for all excitation vectors stored in adaptive codebook 210, and obtains error signal from auditory weighting section 208. When is minimized, the output (P) of the lag parameter encoder 2 15 b, the output (S) of the fixed excitation codebook 2 11, and the output (G) of the gain codebook 2 1 2 Are output to the multiplexer 205. The multiplexer 205 multiplexes the quantized linear prediction coefficients (L) and the outputs (P), (S), and (G), and outputs the multiplexed signals to the modulator 104 of FIG. I do.

The audio decoding unit 110 (FIG. 1) of the wireless communication device is also configured by a general CELP-type audio decoding device, but the description is omitted here.

Next, the lag parameter encoding unit 215 to which the present invention is applied will be specifically described.

FIGS. 3 and 4 show the configuration of a main part of a lag parameter coding unit 215 to which the present invention is applied. FIG. 3 shows a functional block on the lag parameter coding unit side. Indicates a functional block on the lag parameter decoding unit side. Such a lag parameter coding unit is applicable not only to mobile phones but also to all devices that perform voice coding and decoding.

As shown in FIG. 3, the lag parameter coding unit 2 15 includes a lag parameter code book 2 15 a and a lag parameter coding unit that references the lag parameter code book 2 15 a to code the lag value. Means 2 15 b. The lag parameter codebook 2 15 a is a table in which input lag values and output codes are stored in association with each other. It is created under certain rules in advance.

Similarly, as shown in FIG. 4, the lag parameter decoding section of the speech decoding apparatus includes the same lag parameter codebook 2 15 a as the lag parameter coding section, and the lag parameter codebook ₂ 15 Referring to _a, the lag parameter decoding means 4 0 1 to decode the corresponding lag parameter from the received-input code, and a. The lag parameter encoding unit 2 15 having the above configuration will be described more specifically.

The lag parameter codebook 2 15 a is a table in which the relationship between the lag parameter value T and the corresponding code P is represented. For example, if the codebook size is N, the lag value corresponding to the code P (^ ~ is stored. Also, the ITU-T recommendation G.729 (8kbps CS-ACELP) mentioned in the prior art The intermediate code P0 (0 to N-1) is calculated once by using a formula such as equation (1) for the lag parameter encoding in, and the final code P (= 0 to N-1) for P0 You may memorize the correspondence table.

The lag parameter codebook 2 15a of the present invention is obtained by calculating the decoding lag parameter value when a bit error occurs in a code, which is approximately an integer multiple (including 1) of the decoding lag parameter value when there is no bit error. It is characterized in that it is generated so as to increase the rate of occurrence of bit errors with a value or a value that is approximately a fraction of an integer. The creation method will be described later.

The lag parameter is a parameter related to the pitch period included in the audio signal, but a correct lag value may not be obtained due to a bit error or the like. However, the inventor of the present application has found that when the incorrect decoding lag value is a value that is an integral multiple (including 1) of the correct lag value or a value in the vicinity of 1 / integer, the hearing deterioration is relatively small. I found that I could do it. The reason is that the audio signal decoded or synthesized using the incorrect lag value has a frequency component having a correct pitch period as a part of the spectrum as long as the above condition is satisfied.

As described above, according to the first embodiment, even if the lag parameter is erroneously set to a value that is an integral multiple (including 1) of a correct decoded value or a value that is a fraction of an integer, deterioration in auditory perception is small. . „

WO 99/38157

By constructing the lag parameter codebook 2 15a using the characteristic that there is no 8, it is possible to reduce the perceived speech quality degradation when a bit error occurs in the code.

Next, a method of creating the lag parameter codebook used in the present invention will be described. This lag parameter codebook is designed so that the decoding lag parameter value when a bit error occurs in the code is approximately an integer multiple (including 1) of the decoding lag parameter value when there is no bit error, or approximately 1 / The value is set so that the error rate increases.

FIG. 5 shows a processing procedure for creating the lag parameter codebook. First, in step 501, an initial codebook Table (i) (i = 0 to N-1; N is the codebook size) is set. Here, Table (i) represents the decoded value (scalar—value or vector value may be used) for code i. When this codebook is a lag parameter codebook, Table (i) represents the intermediate code P0 for code i as described in the lag parameter codebook 101 in the first embodiment. You may. Furthermore, the correspondence between the code and the decoded value in the initial codebook can be arbitrarily determined. Next, in step 502, in Table (i), for all combinations of (dH MB) codes in which the Hamming distance d H is within a specific number of bits (= MB), the Decoded value distortions are calculated respectively, and the total sum D o is obtained.

Here, the decoded value distortion between codes differs depending on the parameter represented by the code. In general, the Euclidean distance between decoded values or something similar thereto is used. The method of expressing the distortion measure of the lag parameter is one of the features of the present invention. This will be further described in the second embodiment.

Next, in step 503, a code pair i_a, i_b and a codebook Table (i) in which the Hamming distance dH exceeds the specific number of bits MB (dH> MB) is randomly selected. . Then, in step 504, the code pair After exchanging the decoded values, the sum D of decoded value distortions between codes having a Hamming distance within the specific number of bits is calculated.

Next, in step 505, it is determined whether or not the total distortion D in step 504 is less than the previously calculated total distortion Do. If the number has decreased, in step 506, replacement of the decoded value and update of the distortion sum are performed between the code pairs.

Then, in step 507, the convergence of the distortion sum Do is determined, and the operations in steps 503 to 507 are repeated until the distortion sum converges.

If a lag parameter codebook is created by the above processing, decoding between codes within a specific Hamming distance ^: the total sum of distortions can be reduced, so that the decoded value when a bit error occurs The value is close to the correct decoded value in the case where there is no error, and it is possible to suppress the deterioration of the sound quality in audibility.

In particular, by minimizing the sum of distortions between codes having a Hamming distance within a specific number of bits, it is possible to more effectively suppress deterioration when a bit error with a small number of bits occurs. Further, by randomly selecting a code pair i−a, i−b whose Hamming distance exceeds a specific number of bits, it is possible to reduce the sum of distortions more efficiently. Therefore, even if a bit error occurs, it is possible to further suppress the deterioration of audio quality in the sense of hearing.

In step 503, the code pairs a and i_b randomly selected from the codebook Table (i) are limited to those having a Hamming distance exceeding a specific number of bits. Not limited to.

(Embodiment 2)

The second embodiment is implemented on the same hardware and software as the first embodiment. Difference from lag parameter codebook creation method applied to Embodiment 1 Is that the distortion scale has been changed.

The procedure for creating the codebook is the same as that in FIG. 5 shown in the first embodiment. The difference from the first embodiment is that the scale shown in equation (2) is used as the decoded value distortion between codes used in steps 502 and 504. d (fa, fb) = min (wl X d0 (fb, fa), w2 d0 (£ b _f 2 fa), w3 Xd0 (£ b, 3 Xfa) where fa = Fs / Ta (Hz)… (2 ) fb = Fs / Tb (Hz)

fb≥fa

dO (fx, fy) = | fx-fy | / (fxXfy) ^{1 2} Here, T a and T b are the target codes a, i—b the decoding lag value (unit: sample), fa, fl ^ ¾T a, frequency values for Tb and Tb (Hz :), F s is the sampling frequency (Hz), and d (fa, fb) is the decoded value distortion between code pairs.

Equation (2) does not represent the lag parameter value distortion as a mere Euclidean distance. Equation (2) is defined taking into account the difference between one lag value and an integer multiple of the other lag value (wl, w2, and w3 are the distortions from the integer multiple). Weighting constant), and other definitions realizing the same concept can be used.

By using such a distortion measure, the decoded value between codes having a Hamming distance within a specific number of bits becomes a value close to an integral multiple of one of the decoded values. As described above, the lag parameter is a parameter related to the pitch period included in the audio signal, and the decoding lag value is an integer multiple (including 1) or an integer component of the correct lag value due to a bit error or the like. If the audio signal has a value near 1 of the above, the audio signal decoded or synthesized using that value has a frequency component of the correct pitch period as a part of the spectrum, so that the deterioration in auditory perception is compared. You need less.

In addition, it is possible to define a value near such an integral multiple as a small distortion, and By creating a codebook by minimizing distortion only between codes having a Hamming distance within a specific number of bits, the total distortion can be further reduced. Therefore, if the codebook of the lag parameter is created by the above method, even for parameters such as the lag parameter that are sensitive to the deviation of the decoded value due to errors, the deterioration of the auditory quality at the time of bit error can be more effectively reduced. In addition, encoding and decoding can be performed.

As described above, the invention of the lag parameter encoding method according to the present invention is a method for encoding a lag parameter which is an audio signal encoding parameter. A lag parameter codebook that is set to increase the error rate around an integer multiple (including 1) or a fraction of the integer of the decoding lag parameter when there is no bit error in the decoding lag parameter, Is used to encode the lag parameter.

Further, the invention of the lag parameter decoding method according to the present invention provides a lag parameter code that is encoded on the encoding side by the encoding method described in the above aspect, the same as the lag parameter code on the encoding side. It is decrypted using a book.

In this way, the decoding lag parameter value when a bit error occurs in the code increases the error rate when the decoding lag parameter value is an integer multiple (including 1) or around the integral value of 1 when there is no bit error. By using a codebook that is set to cause the lag parameter to have a characteristic that there is little deterioration in auditory perception even if the lag parameter is erroneous near an integer multiple (including 1) or a fraction of the correct decoded value. By using this, it is possible to suppress the deterioration of audio quality in audibility when a bit error occurs in the code.

Further, the invention of the codebook creating method according to the present invention is such that the sum in the codebook of the decoded value distortion between codes having a Hamming distance within a predetermined number of bits in the codebook is set to a minimum or a value close to the minimum. This is a method of creating a codebook. When a bit error occurs in the code, the decoded value is set to a value close to the correct decoded value when there is no error. By restricting the target to codes between Hamming distances within a specific number of bits, Deterioration of sound quality when a bit error occurs in the number of bits can be suppressed more effectively.

In addition, when creating the codebook, a step of calculating the sum of decoded value distortions between codes having a Hamming distance within a predetermined number of bits in the initial codebook, and randomly selecting a code pair from the codebook Calculating the sum of the decoded value distortions between codes having a Hamming distance within the predetermined number of bits after exchanging the decoded value between the code pairs; and When the sum of the distortions is smaller than the previously calculated sum of the distortions, a step of exchanging the decoded values and updating the sum of the distortions; and a step of determining the convergence of the sum of the distortions. The steps of randomly selecting the code pair, exchanging decoded values and updating the distortion sum, and determining the convergence of the distortion sum may be repeated.

In addition, it is preferable to use a distortion criterion such that the distortion is evaluated to be small at a value that is an integral multiple or a fraction of the decoding lag parameter value in determining the decoded value distortion between the lag parameter codes. Using the characteristic that even if an error is near an integer multiple (including 1) of the correct decoded value or a value that is a fraction of an integer, there is little deterioration in auditory perception. Audio quality degradation can be suppressed.

In addition, lag parameter encoding and decoding for encoding and decoding lag parameters by using the above encoding method and decoding method, or using a codebook created by any of the above codebook creating methods. The method can be realized.

Further, the present invention includes a codebook representing a correspondence between a parameter value of a lag parameter, which is a coding parameter of an audio signal, and a code, and a lag parameter encoder for coding the lag parameter using the codebook. Also, it can be realized as a speech encoding device. Further, the present invention can also be realized as a speech decoding device including a lag parameter decoder that decodes the lag parameter code encoded by the encoding device using the same codebook as the encoding side. In addition, The data encoding / decoding device can be realized by one device.

Further, the above-mentioned encoding method can be realized by computer software. More specifically, a computer readable medium and a decoding lag parameter value when the decoding lag parameter value has no bit error when a bit error occurs in the code of the lag parameter which is an encoding parameter of the audio signal. A lag parameter set to be erroneous in the vicinity of an integer multiple (including 1) or a fraction of an integer. A program instruction means for causing a computer processor to encode a lag parameter using a codebook is provided. The program instructing means may be stored in an executable form on the medium, and may be operated by a computer when being executed by the processor.

Similarly, it is of course possible to realize the above-mentioned decoding method by computer software.

Further, the encoding software can be stored in various storage media and used. That is, when a bit error occurs in the code of the lag parameter, which is the coding parameter of the audio signal, the decoding lag parameter value is an integer multiple (including 1) of the decoding lag parameter value when there is no bit error. Alternatively, it is a machine-readable storage medium that stores a program that encodes a lag parameter using a lag parameter codebook that is set to be erroneous near a value of 1 in an integer. Then, it is downloaded to a computer, and the above encoding method is realized by operating the computer.

Similarly, it is of course possible to store the decryption software in various storage media and use it.

In addition, the computer and the processor must have a codebook so that the sum of the decoded value distortion between the code readable by the computer and the code having a Hamming distance within a predetermined number of bits in the codebook is a minimum or a value close to the minimum. And a program instruction means for creating the program. It can be realized as a codebook creating device that is loaded into a computer memory when the processor executes the codebook and runs a computer.

The present invention can be applied to a case where no error detection is performed. However, it is needless to say that the present invention can be used in combination with the error detection. Further, the present invention can be applied to all voice coding / decoding methods that perform lag parameter coding.

This application is based on Japanese Patent Application No. 2003-93332 filed on Jan. 27, 1998. All its contents are included here. Industrial applicability

INDUSTRIAL APPLICABILITY The encoding device, the decoding device, the encoding method, and the decoding method according to the present invention can be widely applied to devices having a speech encoding device and a speech decoding device. In particular, it is preferable to apply the present invention to a wireless communication device such as a digital mobile phone, because it is possible to efficiently suppress deterioration in auditory quality.

Claims

The scope of the claims

1. A method of encoding a lag parameter, which is an encoding parameter of an audio signal, where the decoding lag parameter value is an integer when there is no bit error when a bit error occurs in the lag parameter code. A lag parameter encoding method characterized by encoding a lag parameter using a lag parameter codebook that is set to be erroneous near a multiple (including 1) or a value that is a fraction of an integer. .

2. The encoding of the lag parameter encoded by the encoding method according to claim 1 on the encoding side by using the same lag parameter codebook as the encoding side. Encoding method.

3. A codebook creating method characterized in that a codebook is created so that the total sum of decoded value distortions between codes having a Hamming distance within a predetermined number of bits in the codebook is a minimum or a value close to the minimum.

4. In the initial codebook, a step of calculating a sum of decoded value distortions between codes whose Hamming distance is within a predetermined number of bits, a step of randomly selecting a code pair from the codebook, Calculating the sum of the decoded value distortions between codes having a Hamming distance within the predetermined number of bits after exchanging the decoded values; and if the sum of the decoded value distortions is smaller than the previously calculated sum of the distortions. Replacing the decoded values and updating the sum of distortions; and determining the convergence of the sum of distortions.

4. The method according to claim 3, wherein the steps of randomly selecting the code pair, exchanging decoding values and updating the distortion sum, and determining the convergence of the distortion sum are repeated until the distortion sum converges. Codebook creation method described.

5. The decoded value distortion between codes, wherein a distortion scale is used such that the distortion is evaluated to be small at an integral multiple of the decoding lag parameter value or at a value of 1 of an integer. Codebook creation method. The codebook creation method according to claim 3.

6. The codebook creation according to claim 4, characterized in that a distortion measure is used such that the distortion is evaluated to be small at an integral multiple of the decoding lag parameter value or 1 / integer as the decoding value distortion between codes. Method.

7. A lag parameter encoding / decoding method characterized by performing lag parameter encoding / decoding by the encoding method according to claim 1 and the decoding method according to claim 2.

8. A lag parameter encoding / decoding method, characterized in that lag parameters are encoded and decoded using the code book created by the code book creating method according to claim 3.

9. A lag parameter encoding / decoding method, characterized in that lag parameters are encoded / decoded using the codebook created by the codebook creating method according to claim 4.

10. A lag parameter encoding / decoding method characterized in that lag parameters are encoded / decoded using the codebook created by the codebook creating method according to claim 5.

11. A lag parameter encoding / decoding method characterized by performing lag parameter encoding / decoding using the codebook created by the codebook creating method according to claim 6.

12. A codebook representing a correspondence between a parameter value of a lag parameter, which is a coding parameter of an audio signal, and a code, and a lag parameter encoder for coding a lag parameter using the codebook, wherein the code The ratio of the error in the book when the decoding lag parameter value when a bit error occurs in the code is around an integral multiple (including 1) or 1 / integer value of the decoding lag parameter value when there is no bit error. A lag parameter encoding device characterized by being set to increase.

13 3. On the encoding side, a code encoded by the encoding device according to claim 12 A lag parameter decoding device equipped with a lag parameter decoder that decodes the code of the lag parameter using the same codebook as the encoding side.

14. A lag parameter coding / decoding device that performs lag parameter coding / decoding by the coding device according to claim 12 and the decoding device according to claim 13.

15. An audio encoding / decoding device comprising the lag parameter encoding / decoding device according to claim 14.

16. A wireless communication device comprising the speech encoding / decoding device according to claim 15.

17. Computer-readable media and decoding lag parameters when a bit error occurs in the lag parameter code, which is an audio signal coding parameter.

The computer uses a lag parameter codebook that is set to be erroneous near an integer multiple (including 1) or a fraction of the integer value of the decoding lag parameter when there is no bit error. A program instruction means for causing a processor to encode a lag parameter,

An encoding device in which the program instruction means is stored in an executable form on the medium, and is loaded into a computer memory to move the computer when the processor is executed.

1 8. Decoding lag parameter value when there is no bit error when a bit error occurs in the medium that can be read by a computer and the lag parameter code that is the coding parameter of the audio signal. A program instruction means for causing a computer processor to decode the lag parameter code using a lag parameter codebook set to be erroneous near an integral multiple (including 1) or a fraction of an integer Have,

A decoding device in which the program instruction means is stored in an executable form on the medium, and is loaded into a computer memory to move a computer when the processor executes the program instruction means.

19. The codebook is sent to the computer processor so that the sum of the decoded value distortions between the code readable by the computer and the code having the Hamming distance within a predetermined number of bits in the codebook is a minimum or a value close to the minimum. And program instruction means for creating

A codebook creating apparatus in which the program instruction means is stored in an executable form on the medium, and is loaded into a computer memory to run a computer when executed by the processor.