Speech coding system to reduce distortion through signal overlap

An adaptive codebook having excitation signal predetermined in the past, an excitation codebook for vector quantizing an excitation signal of the input speech signal and a gain codebook for vector quantizing gains of the adaptive and excitation codebooks are provided. A perceptually weighted speech signal having a subframe length obtained by dividing the frame is developed by using the input speech signal and the spectral parameters. A zero input signal of a synthesis filter is developed for a predetermined length by providing the input speech signal of the present subframe as an initial value to the synthesis filter on the basis of the spectral parameters. An overlap signal is also developed by weighting the zero input signal on the basis of the spectral parameters. Optimal codevectors are searched from the adaptive, excitation and gain codebooks according to a signal obtained by connecting the overlap signal to the trailing end of the perceptually weighted speech signal.

BACKGROUND OF THE INVENTION 
The present invention relates to a speech coding system for high quality 
coding speech signals at a low bit rate, particularly a bit rate of 8 
kb/sec or less, with a comparatively small amount of operations. 
As a prior art speech coding system for vector quantizing an excitation 
signal with an excitation codebook, a CELP system is well known. This 
system is disclosed in a treatise by M. R. Shroeder and B. S. Atal 
entitled "Code-Excited Linear Prediction (CELP): High-Quality Speech at 
Very Low Bit Rates", Proc. ICASSP for Acoustic, Speech and Signal 
Processing, 1985, p--p 937-940 (literature 1). Also, as a CELP system 
having an adaptive codebook, a CELP system is well known, which is 
disclosed in a treatise by W. B. Kleijn et al entitled "Improved Speech 
Quality and Efficient Vector Quantization in SELP", Proc. ICASSP for 
Acoustic, Speech and Signal Processing, 1988, p--p 155-158 (literature 2). 
In these CELP systems, optimal codevectors are searched from excitation, 
adaptive and gain codebooks to minimize the perceptually weighted square 
distance between the input and coded speech signals for each subframe 
length. However, since the coding is done for each subframe, distortion is 
liable to result at the block boundary in the block coding, and therefore 
sufficiently satisfactory speech sound quality can not be obtained. To 
alleviate the distortion at the block boundary of the block coding, a 
speech coding system has been proposed in a treatise by LeBlanc et al 
entitled "Structured Codebook Design in CELP". International Mobile 
Satellite Conference, 1990, p--p 667-672 (literature 3). In this system, 
an optimal codevector is searched from an excitation codebook to minimize 
the perceptually weighted square distance between two signals. The first 
signal is obtained by connecting the next subframe input speech signal for 
a predetermined length called overlap length to the present subframe input 
speech signal. The second signal, is obtained by connecting an influence 
signal of a coded speech signal having a length corresponding to the 
overlap length to the trailing end of the coded speech signal. 
In the prior art systems noted above, the distortion at the block boundary 
of the block coding still cannot be sufficiently reduced although the 
distortion can be reduced to a certain degree. 
SUMMARY OF THE INVENTION 
An object of the present invention is therefore to provide a speech coding 
system capable of solving the above problem and obtaining satisfactory 
speech sound quality compared with that in the prior art at a bit rate of 
8 kb/sec or less with a comparatively small amount of operations. 
According to the present invention, there is provided a speech coding 
system comprising a linear prediction analysis section for developing 
spectral parameters of an input speech signal divided at a predetermined 
interval in each frame, an adaptive codebook having excitation signals 
predetermined in the past, an excitation codebook for vector quantizing an 
excitation signal of the input speech signal, a gain codebook for vector 
quantizing gains of the adaptive and excitation codebooks, and a synthesis 
filter for producing a synthetic signal. In this arrangement a 
perceptually weighted speech signal having a subframe length obtained by 
dividing the frame is developed by using the input speech signal and the 
spectral parameters, a zero input signal of a synthesis filter is 
developed for a predetermined length by providing the input speech signal 
of the present subframe as an initial value to the synthesis filter on the 
basis of the spectral parameters, and an overlap signal is developed by 
weighting the zero input signal on the basis of the spectral parameters, 
and optimal codevectors are searched from the adaptive, excitation and 
gain codebooks according to a signal obtained by connecting the overlap 
signal to the trailing end of the perceptually weighted speech signal. 
In another aspect of the present invention, there is provided a speech 
coding system comprising: a linear prediction analysis means for executing 
linear prediction analysis on each subframe of an input speech signal to 
produce LPC coefficient sets; a spectral parameter quantizer means for 
quantizing the spectral parameters corresponding to the LPC coefficient 
sets, and for converting the quantized spectral parameters into LPC 
coefficient sets; a first weighting filter means for executing a 
perceptual weighting of the subframe speech signal on the basis of the 
non-quantized LPC coefficient set of the present subframe supplied from 
the linear prediction analysis means; a synthesis filter means for 
producing a synthetic signal for a predetermined overlap length by setting 
the input speech signal of the present subframe speech signal as an 
initial value, and for setting the excitation signal to zero on the basis 
of the non-quantized LPC coefficient set of the next subframe speech 
signal; a second weighting filter means for weighting the synthetic signal 
on the basis of the non-quantized LPC coefficient set of the next subframe 
supplied from the linear prediction analysis means; a connection circuit 
means for connecting the signal output from the second weighting filter 
means to a trailing end of the signal supplied from the first weighting 
filter means; an influence signal subtraction circuit means for developing 
an influence signal from the previous subframe on the basis of the 
quantized LPC coefficient sets of the present and next subframes supplied 
from the spectral parameter quantizer means, weighting the influence 
signal on the basis of the non-quantized LPC coefficient sets of the 
present and next subframes supplied from the linear prediction analysis 
means to obtain a weighted influence signal, and subtracting the weighted 
influence signal from the output signal from the connection circuit means; 
an adaptive codebook search means for searching for an optimal adaptive 
codevector from an adaptive codebook on the basis of the signal supplied 
from the influence signal subtraction circuit means, the non-quantized LPC 
coefficient sets of the present and next subframes supplied from the 
linear prediction means, the quantized LPC coefficient sets of the present 
and next subframes supplied from the spectral parameter quantizer means 
and an adaptive codevector supplied from the adaptive codebook; and an 
excitation codebook search means for searching for an optimal excitation 
codevector from an excitation codevector on the basis of the signal 
supplied from the influence signal subtraction means, the non-quantized 
LPC coefficient sets of the present and next subframes supplied from the 
linear prediction analysis means, the quantized LPC coefficient sets of 
the present and next subframes supplied from the spectral parameter 
quantizer means, the selected adaptive codevector supplied from the 
adaptive codevector search means and excitation codevector supplied from 
the excitation codebook, and supplying the searched excitation codevector 
to the gain codebook search means and also supplying an index of the 
searched excitation codevector to a multiplexer means. 
Other objects and features will be clarified from the following description 
with reference to attached drawings.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS 
A principle of the speech coding system according to the present invention 
will be described. 
An input speech signal x which is divided into subframes, is weighted by 
the perceptual weighting filter W using a non-quantized LPC (Linear 
Prediction Coding) coefficient set of the present subframe to produce a 
weighted input speech signal x.sub.w. 
The perceptual weighting filter W has a transfer function W(z) given as the 
following formula (1), 
##EQU1## 
In this formula, .alpha..sub.i is a non-quantized LPC coefficient set of 
the present subframe, .beta. and .gamma. are weighting coefficients, and p 
is an order of LPC. 
Using the input speech signal of the present subframe as an initial value, 
a zero input response of a synthesis filter S' using the non-quantized LPC 
coefficient set of the next subframe is developed for the length of 
overlap length L.sub.O. An overlap signal v is then produced by weighting 
with the perceptual weighting filter W' using the non-quantized LPC 
coefficient set of the next subframe. When the present subframe is the 
final subframe, the non-quantized LPC coefficient set of the present 
subframe is used in lieu of the non-quantized LPC coefficient set of the 
next subframe. 
The overlap signal disclosed in the literature 3, is the input speech 
signal of the next subframe. However, according to the present invention 
the signal, which is to be represented by the adaptive, excitation and 
gain codevectors of the present subframe, is an influence signal on the 
next subframe that is based on the present subframe input speech signal. 
Thus, for efficient reduction of the distortion at the block boundary of 
the block coding, generated as a result of coding for each subframe, it is 
preferred to adopt an influence signal on the next subframe based on the 
present subframe input speech signal as the overlap signal. 
The overlap signal v is connected to the trailing end of the weighted input 
signal x.sub.w to produce a signal x.sub.e called an expanded weighted 
input signal. 
With the previous subframe signal as an initial value, the zero input 
response of the synthesis filter S, using the non-quantized coefficient 
set of the present subframe, is obtained for the length of the subframe 
length L.sub.s. With the signal thus obtained as an initial value, the 
zero input response of the synthesis filter S' using the quantized LPC 
coefficient set of the next subframe is obtained for the length of the 
overlap length L.sub.O. Further, the subframe length portion is weighted 
with the perceptual weighting filter W using the non-quantized LPC 
coefficient set of the present subframe, while the overlap length portion 
is weighted with the perceptual weighting filter W' using the 
non-quantized LPC coefficient set of the next subframe, thus obtaining a 
weighted influence signal f. The weighted influence signal f is subtracted 
from the expanded weighted input signal x.sub.e. The signal obtained by 
subtracting the weighted influence signal f from the expanded weighted 
input signal x.sub.e is referred to as signal y. If the present subframe 
is the final subframe, the non-quantized LPC coefficient set of the 
present subframe is used in lieu of the non-quantized LPC coefficient set 
of the next subframe, while using the quantized LPC coefficient set of the 
present subframe in lieu of the quantized LPC coefficient set of the next 
subframe. 
First, an adaptive codevector which can minimize the error E.sub.a in 
formula (2) is searched. 
##EQU2## 
where, 
##EQU3## 
In the formula, sa.sub.d is a perceptually weighted synthetic signal, which 
is obtained with the synthesis filters S and S' and perceptual weighting 
filters W and W' from an expanded adaptive codevector a.sub.d obtained by 
providing L.sub.O "O"s in succession after an adaptive codevector having a 
delay d, and g.sub.a is an optimum gain of the perceptually weighted 
synthetic signal of the expanded adaptive codevector a.sub.d. 
The optimum gain g.sub.a of the perceptually weighted synthetic signal 
sa.sub.d of the expanded adaptive codevector a.sub.d is given as: 
##EQU4## 
By substituting this formula into formula (2), the following formula is 
obtained: 
##EQU5## 
where, 
##EQU6## 
Next, an excitation codevector which can minimize the error E.sub.e in the 
following formula (7) with respect to the selected adaptive codevector is 
searched. 
EQU E.sub.e =.parallel.y-g.sub..alpha. s.alpha..sub.d -g.sub.e 
se.sub.i.sup..perp. .parallel..sub.L.alpha.+Lo.sup.2 (7) 
In this formula, se.sub.i.sup..perp. is an orthogonalized perceptually 
weighted synthetic signal of expanded excitation codevector e.sub.i, which 
is obtained by orthogonalizing the perceptually weighted synthetic signal 
se.sub.i which is obtained with the synthesis filters S, S' and perceptual 
weighting filters W, W' from the expanded excitation codevector e.sub.i 
produced by providing L.sub.0 "O"s in succession after the excitation 
codevector of index i, with respect to the perceptually weighted synthetic 
signal sa.sub.d of the selected expanded adaptive codevector sa.sub.d, and 
g.sub.e is the optimum gain of the orthogonalized perceptually weighted 
synthetic signal se.sub.i.sup..perp.. The gain g.sub.e is given by the 
following formula (8). 
##EQU7## 
This formula is substituted into the formula (7) to develop the following 
formulae: 
##EQU8## 
where, 
##EQU9## 
Finally, a gain codevector which can minimize the error E.sub.g in the 
following formula (12), is searched with respect to the selected expanded 
adaptive and excitation codevectors a.sub.d and e.sub.i. 
EQU E.sub.g =.parallel.y-G1.sub.k s.alpha..sub.d -G2.sub.k se.sub.i 
.parallel..sub.Ls +Lo.sup.2 (12) 
Here, (G1.sub.k, G2.sub.k) is the gain codevector of index k. 
As the vector (G1.sub.k, G2.sub.k) may be used, instead of the gain 
codevector itself, a gain codevector which is obtained through conversion 
of a matrix calculated by using, for instance, a quantized power of the 
weighted input signal, a power of residual signal estimated from an LPC 
coefficient set, powers of the expanded adaptive and excitation 
codevectors. 
Now, in the following description, when a present subframe is the final 
subframe, the term "non-quantized LPC (linear prediction coding) 
coefficient set of the next subframe" refers to the non-quantized LPC 
coefficient set of the present sub-frame, and the term "quantized LPC 
coefficient of the next subframe" refers to the quantized LPC coefficient 
set of the present subframe. 
Referring to FIG. 1 a speech signal which has been divided for each frame 
(for instance of 40 msec.), which appears at an input terminal 1, is fed 
to a linear prediction analysis circuit 2 and also to a subframe division 
circuit 3. 
The linear prediction analysis circuit 2 performs linear prediction 
analysis of the input speech signal, and supplies obtained spectral 
parameter to a weighting filter 4, a synthesis filter 14 and a weighting 
filter 15 in an overlap signal generation circuit 5, an influence signal 
subtraction circuit 6, an adaptive codebook search circuit 7, an 
excitation codebook search circuit 8, a gain codebook search circuit 9, 
and a spectral parameter quantizer 17. 
The spectral parameter quantizer 17 converts the LPC coefficient set 
supplied from the linear prediction analysis circuit 2 into a spectral 
parameter to be quantized (but does not convert when quantizing the LPC 
coefficient set itself), and quantizes the spectral parameter (by 
converting the LPC coefficient set into a LSP (line spectrum pair) set and 
then vector-scalar quantizing the LSP set, for instance). Then, the 
spectral parameter quantizer 17 converts the spectral parameter obtained 
by the quantization into an LPC coefficient set and supplies the LPC 
coefficient set thus obtained to the influence signal subtraction circuit 
6, and adaptive, excitation and gain codebook search circuits 7, 8 and 9. 
Further, an index of the quantized spectral parameter is supplied to a 
multiplexer 13. 
The weighting filter 4, receives from the subframe division circuit 3, the 
input speech signal divided into the subframe length (of 8 msec., for 
instance), executes perceptual weighting of the input speech signal of the 
subframe length in accordance with formula (1) by using the non-quantized 
LPC coefficient set of the present subframe input from the linear 
prediction analysis circuit 2, and feeds the data thus obtained to the 
connection circuit 16. 
The synthesis filter 14 produces a synthetic signal for the overlap length 
with the input speech signal of the present subframe input from the 
subframe division circuit 3 as an initial value, with the excitation 
signal set to zero, and using the non-quantized LPC coefficient set of the 
next subframe input from the linear prediction analysis circuit 2, and 
feeds the synthetic signal to the weighting filter 15. 
The weighting filter 15 executes weighting of the input signal from the 
synthesis filter 14 in accordance with formula (1) by using the 
non-quantized LPC coefficient set of the next subframe supplied from the 
linear prediction analysis circuit 2, and supplies the weighted input 
signal to the connection circuit 16. Here, it is possible to alternatively 
use the quantized LPC coefficient set supplied from the spectral parameter 
quantizer 17 in lieu of the non-quantized LPC coefficient set. 
The connection circuit 16 connects the signal supplied from the weighting 
filter 15 to the trailing end of the signal supplied from the weighting 
circuit 4, and supplies the resultant signal to the influence signal 
subtraction circuit 6. 
The influence signal subtraction circuit 6 calculates an influence signal 
from the previous subframe by using the quantized LPC coefficient sets of 
the present and next subframes supplied from the spectral parameter 
quantizer 17 and executes weighting by using the non-quantized LPC 
coefficient sets of the present and next subframes supplied from the 
linear prediction analysis circuit 2, thus obtaining a weighted influence 
signal. Then, the influence signal subtraction circuit 6 subtracts the 
weighted influence signal from the signal supplied from the connection 
circuit 16 and supplies the resultant difference signal to the adaptive, 
excitation and gain codebook search circuits 7, 8 and 9. The weighting may 
be executed by using the quantized LPC coefficient set output from the 
spectral parameter quantizer 17 in lieu of the non-quantized LPC 
coefficient set as well. 
The adaptive codebook search circuit 7 calculates an error E.sub.a in 
accordance with formula (5) by using the signal supplied from the 
influence signal subtraction circuit 6, the non-quantized LPC coefficient 
sets of the present and next subframes supplied from the linear prediction 
circuit 2, the quantized LPC coefficient sets of the present and next 
subframes supplied from the spectral parameter quantizer 17 and the 
adaptive codevector supplied from the adaptive codebook 10, and executes 
search of an adaptive codevector which minimizes the error E.sub.a. Thus 
selected adaptive codevector is supplied to the excitation and gain 
codebook search circuits 8 and 9 and the delay d of the selected adaptive 
codevector is supplied to the multiplexer 13. 
The excitation codebook search circuit 8 calculates an error E.sub.e in 
accordance with formulae (9) to (11) by using the signal supplied from the 
influence signal subtraction circuit 6, the non-quantized LPC coefficient 
sets of the present and next subframes supplied from the linear prediction 
analysis circuit 2, the quantized LPC coefficient sets of the present and 
next subframes supplied from the spectral parameter quantizer 17, the 
selected adaptive codevector supplied from the adaptive codevector search 
circuit 7 and excitation codevector supplied from the excitation codebook 
11, and executes search of an excitation codevector which minimizes the 
error E.sub.e. Then, the excitation codebook search circuit 8 supplies the 
excitation codevector thus selected to the gain codebook search circuit 9 
and also supplies an index of the selected excitation codevector to the 
multiplexer 13. To reduce the amount of operations in the calculation of 
E.sub.e, it is possible to obtain an auto-correlation of weighted 
synthetic signal for expanded excitation codevector signal se.sub.i in 
accordance with the following formula (13) on the basis of an 
auto-correlation approximation method, which is disclosed in a treatise by 
M. Trancoso and B. Atal and entitled "Efficient Search Procedures for 
Selecting the Optimum Innovation in Stochastic Coders", IEEE Trans. 
Acoust., Speech, Signal Processing, vol. 38, p--p 385-396 (literature 3). 
##EQU10## 
In this formula, hh is an auto-correlation function of the impulse response 
of a weighting synthesis filter WS, which is formed from a synthesis 
filter S using the quantized LPC coefficient set of the present subframe 
and a weighting filter W using the non-quantized LPC coefficient set of 
the present subframe, ee.sub.i is an auto-correlation function of the 
excitation codevector of index i, and im is the impulse response length. 
To reduce the amount of operations, the cross-correlation between the 
weighted synthetic signal for the expanded excitation codevector se.sub.i 
and a given vector v, may be obtained in accordance with the following 
formula (14). 
EQU &lt;.nu.,se.sub.i &gt;.sub.Ls+Lo =&lt;H.sup.T .nu.,e.sub.i &gt;.sub.Ls (14) 
Here, H is the impulse response matrix of the weighting synthesis filter 
WS, and H.sup.T is the transposed matrix of H. 
It is possible to obtain the cross-correlation between the weighted 
synthetic signal for the expanded adaptive codebook sa.sub.d and a given 
vector v likewise in accordance with the following formula (15). 
EQU &lt;.nu.,s.alpha..sub.d &gt;.sub.Ls+Lo =&lt;H.sup.T .nu.,.alpha..sub.d &gt;.sub.Ls(15) 
The gain codebook search circuit 9 executes search of a gain codevector 
which can minimize the error E.sub.g in accordance with formula (12) by 
using the signal supplied from the influence signal subtraction circuit 6, 
the non-quantized LPC coefficient sets of the present and next subframes 
supplied from the linear prediction analysis circuit 2, the quantized LPC 
coefficient sets of the present and next subframes supplied from the 
spectral parameter quantizer 17, the selected adaptive codevector supplied 
from the adaptive codebook search circuit 7, the selected excitation 
codevector supplied from the excitation codebook search circuit 8 and the 
gain codevector supplied from the gain codebook 12. The gain codebook 
search circuit 9 supplies the gain codevector thus selected to the gain 
codebook search circuit 9 and also supplies an index of the selected gain 
codevector to the multiplexer 13. 
While in this embodiment uses perceptually weighted, non-quantized LPC 
coefficient sets in the adaptive, excitation and gain codebook search 
circuits 7, 8 and 9, it is possible to use, alternatively, the quantized 
LPC coefficient set supplied from the spectral parameter quantizer 17. 
Further, while in this embodiment the same overlap length is set for the 
adaptive, excitation and gain codebook search circuits 7 to 9, it is also 
possible to set different overlap lengths for these circuits. 
As has been described in the foregoing, in the system according to the 
present invention, to search the adaptive, excitation and gain codebooks, 
a perceptually weighted signal having the subframe length is obtained by 
using an input speech signal and spectral parameter determined as a result 
of the linear prediction analysis of the input speech signal, an overlap 
signal having a predetermined length is obtained by using the perceptually 
weighted signal and spectral parameter, and the adaptive, excitation and 
gain codebooks are searched by using a signal obtained by connecting the 
overlap signal to the trailing end of the perceptually weighted signal. 
As a result, the speech signal that is represented by the adaptive, 
excitation and gain codevectors of the present subframe consists of the 
input speech signal of the present subframe and a influence signal based 
on the present subframe input speech signal and a non quantized LPC 
coefficient set of the next subframe. In addition by using an influence 
signal of the present subframe input speech signal on the next subframe, 
the distortion of the block boundary of block coding that is generated by 
coding for each subframe, can be reduced more effectively than in the 
prior art system using the next subframe input speech signal as the 
overlap signal (i.e., system disclosed in literature 3).