A speech coding/decoding system

Abstract: An input speech signal is encoded by an adaptive quantizer (16) which quantizes the predicted residual signal produced by removing correlations from the digital input signal by predictors (6, 10), wherein a coefficient or weighting factor called a leakage at the predictor (6) is adaptively adjusted by a leakage selector (47) depending upon a prediction gain which indicates the accuracy of the prediction. The value of leakage is in the range between 0 and 1, depending upon the speech signal is voiced sound or unvoiced sound.

Description: BACKGROUND OF THE INVENTION The present invention relates to a speech signal coding/decoding system for coding/decoding a digital input speech signal at a low bit rate. In a system with a restricted frequency bandwidth and/or transmission power, such as a digital maritime satellite communication system or a digital business satellite communication system employing an SCPC (single channel per carrier), the speech coding/decoding system which can achieve a high speech quality at low bit rate and is little affected by a transmitted code error is required. Based on such a background, a variety of the speech coding/decoding systems have been already proposed. The typical systems thus proposed include an adaptive predictive coding (APC) system for coding an input signal on a frame basis with a predictor for removing a correlation from the input signal in order to obtain a residual signal, and an adaptive quantizer for quantizing the residual signal (USP 4,811,396,), a multi-pulse excited linear predictive coding (MPEC) system for exciting an LPC synthetic filter by a plurality of pulses as a sound source and a CELP (code excited linear predictive coding) system for exciting an LPC synthetic filter by a residual signal pattern as the sound source, and the like. The adaptive predictive coding (APC) system will be described below in detail as the typical example of a conventional speech coding/decoding system. Figs.1(a) and 1(b) show the fundamental structure of a conventional adaptive predictive coding system (USP 4,811,396). In operation, a digital input signal is inputted to an LPC analyzer 2 and a short term predictor 6 via a coder input terminal 1. A short term spectral analysis (called "LPC analysis" hereinafter) is conducted on every frame by the LPC analyzer 2 based on the digital input signal. An LPC parameter obtained thereby is coded by an LPC parameter coder 3 to be transmitted to a decoder on a receiving side via a multiplexer 30. The output of the LPC parameter coder 3 is decoded by an LPC parameter decoder 4. A short term prediction parameter is obtained from the output of the decoder 4 by an LPC parameter/short term prediction parameter converter 5. The short term prediction parameter is set to a short term predictor 6, a noise shaping filter 19 and a local decoding short term predictor 24. A correlation between the adjacent samples of a speech waveform is removed by subtracting the output of the short term predictor 6 employing the short term prediction parameter from the digital input signal by a subtracter 11 to obtain a short term prediction residual siganl. This signal is inputted to a pitch analyser 7 and a long term predictor 10. A pitch analysis is conducted on every frame by the pitch analyzer 7 based on the short term prediction residual signal. A pitch period and a pitch parameter obtained thereby are coded by a pitch parameter coder 8 to be transmitted to the decoder on the receiving side via the multiplexer 30. On the other hand, the pitch period and the pitch parameter are decoded by a pitch parameter decoder 9 to be set to a long term predictor 10, the noise shaping filter 19 and a local decoding long term predictor 23. The periodicity of the short term predictor signal is removed by subtracting the output of the long term predictor 10 employing the pitch period and the pitch parameter from the short term prediction residual signal by a subtracter 12 to obtain a long term prediction residual signal which is ideally white noise. The output of the noise shaping filter 19 is subtracted from the long term prediction residual signal by a subtracter 17 to obtain a final prediction residual signal. This signal is quantized and coded by an adaptive quantizer 16 to be transmitted to the decoder on the receiving side via the multiplexer 30. The coded final predicted residual signal is decoded and inversely quantized by an inverse quantizer 18 to be inputted to a subtracter 20 and an adder 21. A quantization noise is obtained by subtracting the final predicted residual signal, an input signal to the adaptive quantizer 16, from the inversely quantized final predicted residual signal. The quantization noise is inputted to the noise shaping filter 19. In order to update a step size of the adaptive quantizer for every subframe, an RMS (root mean square) value of above-described long term predicted residual signal is calculated by an RMS value calculating circuit 13 to be coded as a reference level by an RMS value coder 14. The RMS value coder 14 stores a reference level and adjacent levels. The output signal of the RMS value coder 14 is decoded by an RMS value decoder 15 and a quantized RMS value corresponding to the reference level in particular is made as a reference RMS value. The step size of the adaptive quantizer 16 is determined by multiplying the reference RMS value by a fundamental step size prepared in advance. On the other hand, the output of the local decoding long term predictor 23 is added to a quantized final predicted residual signal, the output signal of the inverse quantizer 18, by the adder 21. An obtained resultant is inputted to the local decoding long term predictor 23 and added thereto with the output of the local decoding short term predictor 24 by an adder 22 to be inputted to the local decoding short term predictor 24. A locally decoded digital input signal is obtained by such a procedure. A difference between the locally decoded digital input signal and the original digital input signal is obtained as an error signal by a subtracter 26. The power of the error signal is calculated by a minimum error power detector 27 over the sub-frames. A series of similar operations are performed with respect to other fundamental step sizes prepared in advance and the stored adjacent levels to the reference level. The coded RMS level and the fundamental step size that provide the minimum power in error signal powers thus obtained are selected to be transmitted to the decoder on the receiving side via the multiplexer 30. A step size coder 29 is employed for coding the step size. Fig.1(b) is a block diagram showing the decoder employed in a conventional adaptive predictive coding system. Codes inputted via a decoder input terminal 32 are separated into signals relating to a final residual signal, the RMS value, the step size, the LPC parameter, the pitch period and the pitch parameter by a demultiplexer 33 to be inputted to an adaptive inverse quantizer 36, an RMS value decoder 35, a step size decoder 34, an LPC parameter decoder 38 and a pitch parameter decoder 37, respectively. The RMS value decoded by the RMS value decoder 35 and the fundamental step size obtained by the step size decoder 34 are set to the adaptive inverse quantizer 36. A series of codes relating to the received final predicted residual signal is inversely quantized by the adaptive inverse quantizer 36 to obtain a quantized final predicted residual, signal. On the other hand, a short term prediction parameter decoded by the LPC parameter decoder 38 and obtained by an LPC parameter/short term prediction parameter converter 39 is set to the short term predictor 43, one of the predictors which form the synthetic filter, and to a post noise shaping filter 44. The pitch period and the pitch parameter which are decoded by the pitch parameter decoder 37 are set to a long term predictor 42, the other predictor that forms the synthetic filter. The output of the long term predictor 42 is added to the output of the adaptive inverse quantizer 36 by an adder 40. The output thereof is inputted to the long term predictor 42. Further the output of the adder 40 is added to the output of the short term predictor 43 by an adder 41 to obtain a reproduced speech signal. This signal is inputted to the short term predictor 43 and the post noise shaping filter 44 to noise-shaping. Further, the reproduced speech signal is inputted also to a level adjuster 45 and the level is adjusted by comparing the reproduced speech signal with the output of the post noise shaping filter 44. Specifically, a gain adjustment coefficient G₀ is obtained by; and the output of the post noise shaping filter 44 is multiplied by G₀. Then, the short term predictors 6, 24 and 43 in the coder and the decoder will be described below. The transfer function P s (z) of the short time predictors 6, 24 and 43 is given by; where a i is a short term prediction parameter and N s is the number of taps of the short term predictor. The parameter a i is calculated in the LPC analyzer 2 and the LPC parameter/short term prediction parameter converter 5 for every frame and adaptively changes in response to a change in the spectrum of the input signal for every frame. The transfer function represented by an expression (2) is incorporated also into the noise shaping filter 19 in the coder and the post noise shaping 45 in the decoder. Generally, in order to keep the stability of the speech reproduction in the synthetic filters 24 and 43, a prediction obtained by the LPC analyzer 2 is intentionally reduced by introducing a coefficient, called a leakage. That is, generally the product of the leakage r s (0<r s <1) and the short term prediction parameter is employed as a filter parameter for the short term predictors or the noise shaping filters. Specifically, the transfer function P s (z) of the short term predictors 6, 24 and 43 is given by; where the leakage r s is fixed and the same value of the leakage r s is employed on both the coder and decoder sides. The same can be said on the other speech coding/decoding systems. As another example, the CELP system will be described below in brief. On the transmitting side, firstly a correlation between adjacent samples is calculated from the digital input speech signal by the LPC analysis and the short term prediction parameter is set to the synthetic filter. The synthetic filter is excited by a signal outputted from a vector-quantizer to obtain the reproduced speech signal. That is, the short term predicted signal is formed by the short term predictor and added to the exciting signal to reproduce the digital input speech signal in the synthetic filter. The reproduced speech signal is inputted to the short term predictor in order to form the short term predicted signal for the next timing. An error signal between the reproduced speech signal and the digital input speech signal is calculated and the exciting signal is so selected in order to minimize the power of the error signal audibly weighted by the weighting filter. Information on the exciting signal and a short term prediction is transmitted to the receiving side. On the other hand, an exciting signal is formed from the information on the exciting signal by vector-quantizer. Also, on the receiving side as the same as on the transmitting side, the reproduced speech signal is obtained by exciting the synthesis filter with the short term prediction parameter. The short term predictors generally represented by an expression (3) are included in the synthetic filters on the coder side and the decoder side. The leakages are fixed and the same value is employed on both the coder and decoder sides as the same as described above. As described above, such a leakage as the one in the expression (3) is generally employed in the short term predictors 6, 24 and 43, the noise shaping filter 19 and the post noise shaping filter 44. The object of the leakage is to stabilize the operation of the short term predictors 24 and 43, the constituents of the synthetic filter. Conventionally, stability has been attained by intentionally reducing the prediction obtained by the LPC analyzer 2. Therefore, the employment of the small leakage reproduces the speech including much quantization noise especially in the vicinity of a consonant or unvoiced sound. Conversely, the employment of the large leakage reproduces such a speech that appears to resonate especially in the vicinity of a vowel (voiced sound). In the conventional system, however, the constant value leakage has been employed irrespective of the nature of the speech. Therefore, the conventional speech coding/decoding system has the problem that a sufficient decrease in the quantization noise is impossible and a good reproduced speech quality is unable to be obtained in both a voiced sound and an unvoiced sound. SUMMARY OF THE INVENTION It is an object, therefore, of the present invention to overcome the disadvantages and limitations of a prior speech signal coding/decoding system by providing a new and improved speech signal coding/decoding system. It is also an object of the present invention to provide a speech signal coding/decoding system in which the quantization noise is decreased irrespective of a voiced sound and an unvoiced sound, and good speech quality is obtained. The above and other objects are attained by a speech coding/decoding system comprising : a coding side including : a first prediction parameter means for outputting first prediction parameters, an analysis filter comprising a short term predictor for providing a predicted signal of a digital input speech signal based upon said prediction parameters, a quantizer for quantizing a residual signal, said residual signal being obtained by subtracting said predicted signal from said digital input speech signal, a decoding side including : an inverse quantizer for inverse quantization and for decoding of a received quantized residual signal from a transmit side, a second prediction parameter decoder for outputting second prediction parameters, a synthesis filter comprising a short term predictor for reproducing said digital input signal by adding an output of said inverse quantizer and a reproduced predicted signal, characterized by further comprising : in said coding side means for providing the product of a first leakage coefficient (r s ) and said first prediction parameters as parameters for said short-term predictor of said analysis filter, in said decoding side means for providing the product of a second leakage coefficient (r s ) and said second prediction parameters as parameters for said short-term predictor of said synthesis filter ; wherein the value of said leakage coefficients (r s ) is larger than 0 and smaller than 1 and the value of the second leakage coefficient differs from that of the first leakage coefficient. Advantageously, the second leakage coefficient is larger than that of the first leakage coefficient. According to another embodiment of the present invention, the system has a first leakage selector provided in a coding side for adaptively adjusting a coefficient of said predictor based upon said prediction parameter, and a second leakage selector provided in a decoding side for adaptively adjusting a coefficient of said synthesis filter based upon output of said prediction parameter decoder. BRIEF DESCRIPTION OF THE DRAWINGS The foregoing and other objects, features, and attendant advantages of the present invention will be appreciated as the same become better understood by means of the following description and accompanying drawings wherein ; Figs.1(a) and 1(b) are block diagrams of a coder and a decoder, respectively, of a prior speech signal coding/decoding system, Fig.2(a) is a block diagram of a coder according to the present invention, Fig.2(b) is a block diagram of a decoder according to the present invention, Fig.3 is a block diagram of another embodiment of a decoder according to the present invention, and Fig.4 is a block diagram of a decoder of still another embodiment according to the present invention. DESCRIPTION OF THE PREFERRED EMBODIMENTS A first feature of the present invention exists in a constitution wherein a leakage employed in a transmit side and/or a receive side is adaptively adjusted over in accordance with the accuracy of a prediction. A second feature of the present invention is that different values are applied to the leakages employed in a coder and a decoder to code or decode the digital input speech signal. A third feature of the present invention is that the different leakages are employed in the coder and the decoder and a gain difference generated by the different leakages is compensated. Leakages employed in a coder and a decoder and a gain adjustment relating to the leakages which make differences between the present invention and the prior art will be described in detail in a description below. (Embodiment 1) An embodiment 1 has a constitution wherein a leakage employed in a transmit side and/or a receive side is adaptively adjusted over in accordance with the accuracy of a prediction, that is, the leakage in a coder and/or the leakage in a decoder are adaptively changed over. Fig.2(a) shows the constitution of the coder for adaptively adaptively changing over the leakage, which is a first embodiment according to the present invention. A leakage selector 47 (first leakage means) adaptively selects the leakage which is the weighting factor of the predictor by evaluating the accuracy of a prediction by employing an LPC parameter, the output of an LPC parameter decoder 4, to set the leakage to short term predictors 6 and 24 and a noise shaping filter 19. That is, the small leakage is employed in the vicinity of a voiced sound wherein the prediction tends to be right to prevent such a sound as a resonance from being generated and the large leakage is employed in the vicinity of an unvoiced sound wherein the prediction tends not to be right to reduce quantization noise. Thus, a good reproduced speech is obtained by employing the leakage with a suitable magnitude for the nature of a speech. The embodiment according to the present invention is as follows: A kind of prediction accuracy (prediction gain) G p represented by is employed and the leakage r sc is changed over to r sc = r s,1 when G p < G p,th1, and to r sc = r s,2 when G p > G p,th1, (5) where 0< G p,th1 < 1 and 0< r s,1 ≦ r s,2 < 1 The leakage value is fed to the respective short term predictors 6 and 24 and the noise shaping filter 19. Besides changing over the leakage at two steps as described above, the leakage can also be changed over at three steps or more with finer thresholds. A reference r s,1 designates the leakage of a portion wherein the prediction is right, for example, the voiced sound and r s,2 the leakage of a portion wherein the prediction is not right, for example, the unvoiced sound. Fig.2(b) shows the circuit diagram of the docoder in the system according to the present invention. A leakage selector 48 adaptively selects the leakage which is the weighting factor of the synthesis filter by evaluating the prediction accuracy by employing the LPC parameter, the output of the LPC decoder, to set the leakage to the short term predictor 43 and the post noise shaping filter 44. That is, as the same as on a coder side, the small leakage is employed in the vicinity of the voiced sound wherein the prediction tends to be right to prevent such a sound as the resonance from being generated and the large leakage is employed in the vicinity of the unvoiced sound wherein the prediction tends not to be right to reduce the quantization noise. Thus, the good reproduced speech can be obtained by employing the leakage with a suitable magnitude for the nature of the speech. An embodiment of the decoder side is as follows: One of the prediction accuracy given by an expression (4) is employed. The leakage r sd is changed over such that where 0<G p,th2 <1 and 0<r sc r s3 r s4 <1 The leakage value is fed to the short term predictor 43 and the post noise shaping filter 44. Reference r s,3 and r s4 designate the leakages for the voiced sound and the unvoices sound, respectively. Besides changing over the leakage at the two steps of the voiced sound and the unvoiced soundas described above, the leakage can be changed over at three steps or more by employing the finer thresholds. As described above, according to the present invention, the quantization noise can be reduced irrespective of the nature of the speech ; the voice sound or the unvoiced sound, by employing the leakages on the coder and/or decoder sides in accordance with the prediction accuracy. A first leakage selector and a second leakage selector may be implemented by a read only memory. Each address of that memory stores the leakage value depending upon the input signal which is used as an address selection signal of that memory. The input of the LPC parameter decoder 4 in Fig.2(a), or the LPC parameter decoder 38 in Fig.2(b). Those decoders provide the figure indicating the accuracy of the prediction. (Embodiment 2) Next, the second embodiment in which a leakage value in a decoder side differs from a leakage in a coder side is described. As second leakage means, the second feature of the present invention, the larger leakage than that employed on the coder side is set to the short term predictor 43 and the post noise shaping filter 44. The structure of the coder and the decoder are the same as those shown in Figs.1(a) and 1(b), respectively. That is, the second leakage means equivalently improves the predictin accuracy of a short term prediction signal reproduced on the decoder side to reduce the quantization noise. (Embodiment 3) In the embodiment 2, the reproduced speech signal is forced to have a gain due to a difference between the leakages. When the leakages on the coder and decoder sides are different from each other for the purpose of a reduction in the quantization noise, a difference between the gains of the voiced and unvoiced sound portions becomes too distinct due to a difference between the prediction accuracies, coversely resulting in the deterioratin of the speech quality. Thus, in the structure of an embodiment 3, the decoder is provided with a short term predictor 50 for compensating the gain as shown in Fig.3. As the same as in the embodiment 2, the leakage larger than that employed on the coder side is set to the short term predictor 43. The same leakage as that employed on the coder side is set to the gain adjusting short term predictor 50. Further, a short term prediction parameter, the output of the LPC parameter/short term prediction parameter converter 39, is set to the short term predictors 43, 50 and the post noise shaping filter 44. The output signal of the adder 40 is inputted to the adders 41 and 49 and the long term predictor 42. The adder 49 adds the output of the adder 40 and that of the short term predictor 50 to each other and a resultant is inputted to the predictor 50 and the level adjuster 45. On the other hand, the adder 41 adds the output of the short term predictor 43 and that of the adder 40 to each other and a resultant is inputted to the predictor 43 and the post noise shaping filter 44. The output signal of the adder 41 has a gain for the leakage employed in the short term predictor 43 and further has an additional gain by passing the post noise shaping filter. It should be noted that the short term predictor 43 has a leakage which differs from that of the coder side, and the short term predictor 50 has the same leakage as that of the coder side. Therefore, the level of the output of the short term predictor 43 is adjusted by using the output level of the short term predictor 50. The gain is adjusted by the level adjuster 45. Specifically, a gain adjustment coefficient G₀′ is obtained by; from the output of the adder 49 and the output of the post noise shaping filter 44 to be multiplied by the output of the post noise shaping filter 44. Thus, by providing the gain adjusting short term predictor 50, the leakages largely different from each other can be employed on the coder and decoder sides as compared with the embodiment 2, enabling the prediction accuracy to be improved on the decoder side. Therefore, the quantization noise can be resultingly reduced and the speech quality better than that in the embodiment 2 can be obtained. (Embodiment 4) An embodiment 4 has the constitution of the combination of above-described embodiments 1, and 3. A changeover is conducted according to the prediction accuracy and the leakage different from that on the coder side is employed on the decoder side. Fig.4 shows the constitution of the decoder, a fourth embodiment according to the present invention. A leakage selector 51 adaptively selects and sets the leakage for the short term predictor 43, a constituent of the synthetic filter, by evaluating the prediction accuracy by employing the LPC parameter, the output of the LPC parameter decoder 38. The same leakage as that on the coder side is set to a gain adjusting short term predictor 53. The output of the adder 40 is inputted to the long term predictor 42 and the adders 41 and 52. The adder 52 adds the output of the short term predictor 53 and that of the adder 40 to each other and a resultant is inputted to the short term predictor 53 and the level adjuster 45. The embodiment 4 is exemplified as follows: When the prediction accuracy is defined by the expression (4) and the leakage on the coder side is r sc , the leakage r sd on the decoder side is changed over so as to satisfy the following expression: r sd = r sd,1 when G p < G p , th1 and r sd = r sd,2 when G p > G p,th1 , (8) where 0< G p,th1 < 1 and 0< r sc < r sd,1 < r sd,2 < 1 The gain adjustment coefficient G₀ is given by In the embodiment 4, the quantization noise in the whole speech can be reduced by equivalently improving the prediction accuracy of the reproduced short term predicted signal by employing the leakage with a larger value on the decoder side than that on the coder side. Further, the quantization noise can be further decreased by employing the larger leakage in the vicinity of the unvoiced sound wherein the quantization noise tend to be generated than that in the vicinity of the voiced sound. Thus, the reproduced speech quality better than that of above-described embodiments can be obtained in the embodiment 4. As a concrete numerical example, the leakages employed in a hardware with a 9.6 kbps adaptive predictive coding system with maximum likelihood quantization (APC-MLQ) will be mentioned below. o leakage on coder side r sc = 0.9375 o leakage on decoder side r sd = 0.963 when G p < G p,th1, and r sd = 0.973 when G p >G p,th1. While adaptive predictive coding system with the maximum likelihood quantization (APC-MLQ) is exemplified in a description above, the same effect can be obtained by applying the present invention to the other MPEC system, CELP system or the like. As described above, a constitution wherein a coder and a decoder are provided with leakages and the provision of at least one of two leakage means; first leakage means for adaptively changing over the leakages in accordance with the prediction accuracy of a predictive signal and second leakage means for allotting the different leakages determined in advance to a coder side and a decoder side, enable quantization noise to be reduced irrespective of a voiced sound or an unvoiced sound and enable a good reproduced speech quality to be obtained according to the present invention. Since the largely different leakages from each other can be employed on the coder side and the decoder side by providing the second leakage means with gain adjusting means for adjusting the gains of the decoder, the speech quality can be further improved on the decoder side. The provision of the gain adjusting means in addition to the first and second leakage means enables the quantizaion noise to be further reduced irrespective of the voiced sound or the unvoiced sound, and enables the good reproduced speech quality to be obtained. The employment of the LPC parameter for forming the predicted signal enables the excellent prediction accuracy thereof to be realized by the simple constitution without requiring a new circuit. Therefore, a highly efficient speech coding/decoding system at a low bit rate can be obtained according to the present invention and its effect is extremely large. From the foregoing it will now be apparent that a new and improved speech signal coding/decoding system has been found. It should be understood of course that the embodiments disclosed are merely illustrative and are not intended to limit the scope of the invention. Reference should be made to the appended claims, therefore, rather than the specification as indicating the scope of the invention.

Claims: A speech coding/decoding system comprising : a coding side including : a first prediction parameter means (1, 2, 3, 5 ; 7, 8, 9) for outputting first prediction parameters, an analysis filter (6, 10) comprising a short term predictor (6), for providing a predicted signal of a digital input speech signal based upon said prediction parameters, a quantizer (16) for quantizing a residual signal, said residual signal being obtained by subtracting said predicted signal from said digital input speech signal, a decoding side including : an inverse quantizer (36) for inverse quantization and for decoding of a received quantized residual signal from a transmit side, a second prediction parameter decoder (38) for outputting second prediction parameters, a synthesis filter (42, 43) comprising a short term predictor (43) for reproducing said digital input signal by adding an output of said inverse quantizer (36) and a reproduced predicted signal, characterized by further comprising : in said coding side means for providing the product of a first leakage coefficient (r S ) and said first prediction parameters as parameters for said short-term predictor (6) of said analysis filter (6, 10) ; in said decoding side means for providing the product of a second leakage coefficient (r S ) and said second prediction parameters as parameters for said short-term predictor (43) of said synthesis filter (42, 43) ; wherein the value of said leakage coefficients (r S ) is larger than 0 and smaller than 1 and the value of the second leakage coefficient differs from that of the first leakage coefficient. A speech coding/decoding system according to claim 1, wherein the value of the second leakage coefficient is larger than that of the first leakage coefficient. A speech coding/decoding system according to claim 1, further comprising first leakage selector means (47) provided in the coding side for adaptively adjusting said first leakage coefficient depending from prediction gain (G P ) which is based upon said prediction parameters. A speech coding/decoding system according to claim 1, further comprising second leakage selector means (48) provided in the decoding side for adaptively adjusting said second leakage coefficient depending from prediction gain (G P ) which is based upon said decoded prediction parameters. A speech coding/decoding system according to claim 1, wherein a level adjuster (45) is provided on the decoding side, said level adjuster, compensating gain difference between the coding side and the decoding side. A speech coding/decoding system according to claim 1, wherein each of said leakage value is switched between two values depending upon the accuracy of the prediction by the predictor. A speech coding/decoding system according to claim 2, whrein leakage value on the coding side is 0.9375, and leakage value in the decoding side is 0.963 when a prediction gain (G P ) is smaller than a predetermined value and 0.973 when said prediction gain (G P ) is larger than said predetermined value. A speech coding/decoding system according to claim 1, wherein each of said leakage value is selected among more than three values. A speech coding/decoding system according to claim 1, wherein each of said first leakage value and said second leakage value are provided by a ROM.

Inventors: HONMA HIDEKI, IIZUKA SHIGERU, NOMURA TAKAHIRO, YATSUZUKA YOHTARO, HONMA, HIDEKI, IIZUKA, SHIGERU, NOMURA, TAKAHIRO, YATSUZUKA, YOHTARO
Applicants: KOKUSAI DENSHIN DENWA CO LTD, KOKUSAI DENSHIN DENWA CO., LTD
Application year: 1989
Main classification: G10L9
IPC codes: G10L19, H03M3