Digital speech interpolation system

A digital speech interpolation system is combined with an adaptive differential PCM (ADPCM), employing a speech detector for detecting speech signals and for discriminating voiced and unvoiced sounds. An adaptive quantization bit assignment to the speech is adopted to cope with any freeze-out condition. And further PCM speech signals with 8 KHz sampling are applied to ADPCM after shifted 250 Hz down and then converted into 6 KHz sampling frequency, thereby attaining a total gain of about 7 without degrading speech quality.

BACKGROUND OF THE INVENTION 
This invention relates to a digital speech interpolation system, and 
particularly to an efficient digital speech interpolation system in which 
as many digitized speech signals as possible are transmitted through a 
transmission line having a limited communication capacity while avoiding 
freeze-out and assuring a practically satisfactory quality of speech. The 
term "freeze-out" as used throughout the specification and claims is 
intended to describe the condition in which the inputs to the digital 
speech interpolation system overflow the output capacity of such system. 
This invention is useful particularly for a long distance telephone system 
such as international telephone lines, because it can widely improve the 
efficiency of utilization of a transmission line in a digital satellite 
communication system or a digital undersea cable system. 
Owing to highly developed digital signal processing techniques, in 
telephony a digital speech transmission system is practically used in 
which speech signals are digitized to be transmitted. For the purpose of 
economization by efficiently utilizing a transmission line having a 
limited communication capacity, a digital speech interpolation system 
called DSI system is employed in this digital speech transmission system. 
Moreover, a predictive coding system which makes coding with a short bit 
length is used together with the DSI system. In case of transmitting a 
plurality of digital speech signal, the DSI system transmits, by detecting 
sound portions of speech signals in each input trunk, and by combining 
only said detected sound portion to form new digital signals, the new 
digital signals through a smaller number of output channels than the 
number of the input trunks. Generally, digital speech signals are divided 
into unit blocks which are a ground of a predetermined number of serial 
samples. For each unit block, a speech detector detects whether or not 
speech exists in the unit block. The unit blocks in which speech exists 
are transmitted. On the other hand, in the predictive coding system a 
predictor predicts a present sample value from past group of sample values 
of input digital signal. The difference between the predicted value and 
the actual sample value, i.e., prediction error is calculated with a 
subtracter. A quantizer performs quantization of prediction error. By 
abovementioned manner information can be transmitted at a low bit rate. 
Such typical systems include a delta modulation system which performs 
coding with one bit and a differential PCM (DPCM) system which performs 
coding with two or more bits. Among DPCM systems there is an adaptive DPCM 
(ADPCM) system in which the quantization level interval of the quantizer 
and the prediction coefficient of the predictor are controlled so as to be 
of an optimum value at any times. 
An efficient DSI system in which a DSI system is combined with a DPCM 
system or an ADPCM system has been proposed. In the efficient DSI system a 
very high degree of utilization of transmission line is made possible 
owing to the effective utilization of transmission line which is inherent 
in the DSI system transmitting only the speech portions, and owing to the 
band compression in the predictive coding system transmitting the speech 
portions at a low bit rate. Namely, by defining a DSI gain as the ratio of 
the number of DSI input trunks to the number of DSI output channels which 
ratio is determined by the proportion of the detected and transmitted 
speech portions to the whole of input signal on the trunk, and by defining 
a predictive coding gain as a reciprocal of the reduction factor of the 
number of bits after predictive coding to the number of original coding 
bits of speech signal, the total gain of the efficient DSI system may be 
expressed as the product of the DSI gain and the predictive coding gain. 
Although in theory a DSI gain of about 2.5 should be obtained because the 
average operating percentage of speech is generally said to be about 40%, 
in practice the DSI gain is set to about 2 for safety design to avoid 
frequent occurrence of freeze-out. If the DSI gain is set to near 2.5, the 
number of active input trunks of DSI input trunks in which speech is 
existing would tend to instantaneously exceed the number of DSI output 
channels, whereby some of the active input trunks could not be connected 
to an output channel. This would lead to frequent occurrence of freeze-out 
in which speech is not transmitted. On the other hand, in the predictive 
coding system, predictive coding with a fixed length of 4 bits is adopted 
to keep the quality of speech expressed by signal-to quantization noise 
ratio S/N.sub.q at substantially the same degree as normal 8 to 6 bit PCM. 
Only a predictive coding gain of at most 2 can be obtained. In this case, 
the predictive coding maintains redundancy, because 4 bit length coding 
necessary to low S/N.sub.q speech portions is similarly applied to high 
S/N.sub.q speech portions. 
On the other hand, low speed sampling is useful to effectively utilize a 
transmitting line. Really, digitization with 6 KHz sampling has been 
adopted. A 8 KHz sampling is normally adopted for digital speech signal in 
telephony. This is based on the fact that analog speech signals in 
telephony are standardized within a transmission frequency band ranging 
from 0.3 to 3.4 KHz. But, in a speech transmitting system adopting FDM 
(Frequency Division Multiplex) undersea cable transmitting system, 
transmission with 3 KHz band has been practically used, so the 6 KHz 
sampling has been adopted accordingly. More specifically, after speech 
signals transmitted with 8 KHz-8bit PCM are once converted to 8 KHz-13 bit 
linear PCM, they are passed through a low pass filter of 3 KHz band and 
sampling speed is converted to 6 KHz. And the speech signals are 
predictively coded by a 4 bit quantizing and 6 KHz sampling ADPCM encoder 
and transmitted to digital undersea cable or digital satellite 
communication system at 24 kb/s. However, in the speech transmission 
system with 6 KHz sampling the transmitted frequency band is 0.3 to 3.0 
KHz. Therefore, the system has the disadvantage that the high frequency 
components of 3.0 to 3.4 KHz of the speech signals standardized within the 
band of 0.3 to 3.4 KHz are cut off, thereby degrading the quality of 
reproduced speech in its high frequency region. 
SUMMARY OF THE INVENTION 
An object of this invention is to provide an efficient digital speech 
interpolation system wherein avoidance of freeze-out and improvement in 
the DSI gain are attained. Another object of this invention is to provide 
said efficient digital speech interpolation system wherein sampling speed 
is lowered without adversely influencing the transmission frequency band. 
According to an aspect of this invention, in a digital speech interpolation 
system wherein digital input signals are divided into unit blocks which 
are a group of a predetermined number of serial samples and only the unit 
blocks in which speech is existing are transmitted after predictively 
coded, the predictive coding is carried out with a variable quantizing bit 
type predictive encoder, and it is determined whether or not 
signal-to-quantization noise ratio of the speech in the unit blocks in 
which speech is existing is good. If freeze-out occurs in speech 
interpolation, the number of bits in predictive coding is reduced in the 
order of the unit blocks having a better signal-to-quantization noise 
ratio. Preferably, after the frequency is shifted down to a lower one to 
the extent that a considerable degradation of speech does not occur and 
then the sampling speed of the digital input signals is converted to a 
lower one, predictive coding is performed. 
As described above, according to this invention, a predictive encoder with 
a variable bit quantization is employed so that the number of bits in 
predictive coding may be reduced for unit blocks having a good signal-to 
quantization noise ratio when freeze-out occurs. Therefore, the unit 
blocks which would otherwise come under a freeze-out condition can be free 
from freeze-out because of predictive coding with surplus bits by said 
reduction. Furthermore, the unit blocks wherein the number of bits has 
been reduced suffer from no significant degradation of the quality of 
speech because of their good signal-to-quantization noise ratio. For this 
reason, according to this invention, a high DSI gain can be set and an 
efficient utilization of transmission line can be achieved. And, according 
to this invention, since the frequency is shifted down to a lower one with 
the original sampling speed being kept before the conversion of the 
sampling speed into a lower speed, an effective utilization of 
transmission line due to the lowering of the sampling speed can be 
achieved without any considerable degradation of the high frequency 
components of speech.

DETAILED DESCRIPTION OF PREFERRED EMBODIMENTS 
Embodiments of this invention will be described in connection with the 
accompanying drawings. In the embodiments, it is assumed that 8 bit PCM 
signals with 8 KHz sampling are input and unit blocks each formed by 32 
samples thereof are output to DSI output channels after speed conversion 
to 6 KHz sampling. 
The embodiment of this invention shown in FIG. 1 comprises an expander 1, a 
frequency converter 2, a 3 KHz digital low pass filter 3, a sampling speed 
converter 4, a buffer memory 5, a speech detector 6, a variable quantizing 
bit type ADPCM encoder 7 for speech, a data signaling detector 8, an ADPCM 
encoder 9 for data, an assignment controller 10, an assignment status 
signal generator 11, a bit length indicating signal generator 12 and a 
multiplexer 13. A nonlinear PCM signal NL of 8 bits with 8 KHz sampling 
input is converted into a linear PCM signal L.sub.1 of 13 bits with 8 KHz 
sampling by the expander 1. The voice band frequency components of 0.3 to 
3.4 KHz of the linear PCM signal L.sub.1 is shifted 200 Hz down to lower 
range by the frequency converter 2 so as to be of 0.1 to 3.2 KHz. The 
linear PCM signal L.sub.2 including the frequency components of 0.1 to 3.2 
KHz is converted through the digital filter 3 to a linear PCM signal 
L.sub.3 having frequency components of 0.1 to 3.0 KHz which is in turn 
speed converted into a linear PCM signal L.sub.4 with 6 KHz sampling by a 
sampling speed converter 4. In this case, the frequency components of the 
linear PCM signal L.sub.4 are only cut off by 0.2 KHz in higher range 
compared with the linear PCM signal L.sub.1 with 8 KHz sampling. If the 
linear PCM signal L.sub.4 is again shifted 200 Hz up to higher range after 
speed conversion to 8 KHz sampling, a linear PCM signal having frequency 
components of 0.3 to 3.2 KHz is obtained. The linear PCM signal L.sub.4 
with 6 KHz sampling is stored in the buffer memory 5 every 24 samples. 
These 24 samples correspond to unit blocks of 32 samples in the linear PCM 
signal L.sub.1 with 8 KHz sampling. FIG. 2 shows an example of the 
frequency converter 2. This converter 2 comprises four multipliers 2b, 2e, 
2g and 2j, two low pass filters 2d and 2i, and an adder 2l. Signals 
L.sub.5, L.sub.6, L.sub.7 and L.sub.8 are supplied to the input terminals 
2c, 2f, 2h and 2k of the multipliers 2b, 2e, 2g and 2j respectively. These 
signals L.sub.5, L.sub.6, L.sub.7 and L.sub.8 in this example are linear 
PCM signals with 8 KHz sampling which may be expressed by the following 
equation (1). The low pass filters 2d and 2i have a band of 2 KHz. 
##EQU1## 
where 
fo=Wo/2.pi.=2 KHz, 
fx=Wx/2.pi.=fo-.DELTA.fo, and 
.DELTA.fo is a shift value. 
The signal L.sub.1 supplied to the input terminal 2a is multiplied by the 
signal L.sub.5 with the multiplier 2b and passed through the low pass 
filter 2d, and further multiplied by the signal L.sub.6 with the 
multiplier 2e, thereby providing a signal L.sub.9. Similarly, the signal 
.sub.1 is multiplied by the signal L.sub.7 with the multiplier 2g and 
passed through the low pass filter 2i, and further multiplied by the 
signal L.sub.8 with the multiplier 2j, thereby providing a signal 
L.sub.10. The signals L.sub.9 and L.sub.10 are summed with the adder 2l, 
thereby providing a signal L.sub.2 at the output terminal 2m which is 
shifted 200 Hz down. Incidentally, if fx=fo+.DELTA.fo it goes without 
saying that a signal shifted .DELTA.fo up can be obtained. The .DELTA.fo 
may be selected up to 300 Hz. 
While the linear PCM signal L.sub.4 with 6 KHz sampling including speech 
band signals of 0.1 to 3.0 KHz shifted down is stored in the buffer memory 
5, the data and signaling thereof are separated from the input signal and 
encoded wth the ADPCM encoder 9 and transmitted, and only the speech 
portion is sent through the DSI channel CH. The speech portion and data of 
the linear PCM signal L.sub.4 may be processed by the same ADPCM encoder 
without being separated from each other. FIG. 3 shows a portion of FIG. 1 
concerned with speech. The variable quantizing bit type ADPCM encoder 7 
comprises a bit length controller 7a and an ADPCM encoder 7b. The ADPCM 
encoder 7b consists of a subtractor 7c, an adaptive quantizer 7d, and 
adaptive predictor 7e, a dequantizer 7f and an adder 7g. Although the 
block diagram of FIG. 3 shows an independent system, referring to FIG. 1, 
it can be seen that the buffer memory 5 and the speech detector 6 may be 
commonly used for each input trunk TK and that the same number of the 
variable quantizing bit type ADPCM encoders 7 as the number of DSI output 
channels CH are provided. 
The linear PCM signal L.sub.1 with 8 KHz sampling is detected by the speech 
detector 6 every unit block of 32 samples as to whether or not speech is 
existing in the unit block. Moreover, to the unit blocks existing speech 
it is predicted whether or not the signal-to-quantization noise ratio 
S/N.sub.q of the speech is good. To the unit block of 32 samples in which 
the speech detector 6 has detected that speech is existing, a unit block 
of 24 samples of the linear PCM signals L.sub.4 with 6 KHz sampling 
corresponding to this unit block 32 samples is read out from the buffer 
memory 5. The unit block read out is applied to the ADPCM encoder 7b of 
the variable quantizing bit type ADPCM encoder 7. The assignment 
controller 10 controls which variable quantizing bit type ADPCM encoder 7 
be assigned, which belongs to channel CH. Simultaneously, the S/N.sub.q 
good or not information of the read out unit block is sent to the bit 
length controller 7a of the assigned variable quantizing bit type ADPCM 
encoder 7 from the speech detector 6. If freeze-out occurs, the bit length 
controller 7a sends to the quantizer 7d instructions as to the designation 
of the unit block in which the predictive coding bit length should be 
reduced, and as to the bit length to be reduced. In this case, the unit 
blocks in which the bit length should be reduced are designated in the 
order of the unit blocks having better S/N.sub.q. For example, predictive 
coding with normal 4 bit length is made with 3 bit length to the good 
S/N.sub.q unit block. Information as to the bit lengths of each unit block 
is produced by the coding bit length indicating signal generator 12. 
Assignment status signals representing the corresponding relationships 
between the input trunks TK and out put channels CH are produced by the 
assignment status signal generator 11, according to controlling signals 
from the assignment controller 10. 
The speech detector 6 comprises a speech detecting part for detecting 
whether or not speech is existing in the unit block by utilizing an 
internal power in the unit block or the number of zero crossings in the 
unit block, and a speech nature judging part for determining the nature of 
speech in the unit block, that is, whether or not the 
signal-to-quantization noise ratio S/N.sub.q is good. Regarding the 
signal-to-quantization noise ratio S/N.sub.q of speech, generally, the 
speech including many lower frequency components (i.e. voiced sound) has a 
good S/N.sub.q, while the speech including many higher frequency 
components or having flat spectrum (i.e. un-voiced sound) has a poor 
S/N.sub.q. In other words, while most of speeches such as vowel sounds are 
voices and thus have a good S/N.sub.q, un-voices such as fricative 
consonant and a plosive have a poor S/N.sub.q. The speech nature judging 
part of the speech detector 6 may be of the type wherein if it is detected 
that a unit block has extraordinary a number of zero crossings it is 
judged the S/N.sub.q of the unit block is not good, or of the type wherein 
a predictor set to lower frequency range is used and if it is detected 
that in a unit block many wrong predictions occur, it is judged the 
S/N.sub.q of the unit block is poor. In this embodiment a speech detector 
having the structure of circuit as shown in FIG. 4 is used. In FIG. 4, the 
speech detector 6 comprises a power calculator 6b, a zero crossing rate 
counter 6c, a threshold selector 6d, a polarity bit matrix processor 6e 
and a discriminator 6f. For the signal L.sub.1 of 8 KHz.multidot.13 bit 
applied to the input terminal 6a, the power calculator 6b is for 
calculating the internal power St every unit block .tau. of 32 samples. 
The internal power St is compared with a reference value of S.sub.Th by 
the discriminator 6f. When St&lt;S.sub.Th, it is judged that the unit block 
.tau. is silence. On the other hand, when S.sub.Th .ltoreq.St, it is 
judged that the unit block .tau. includes speech, and then a speech 
detecting signal VD(t) is produced from the output terminal 6g. The 
judgement, as to whether or not the S/N.sub.q of the unit block .tau. 
producing the speech detecting signal VD(t) is good, is made as follows: 
Taking notice of a main series which is a series of the polarities of the 
respective sample values in the unit block .tau. itself and a sub-polarity 
series which is a series of the polarities of the respective sample values 
in the unit block .tau. passed through the digital filter, whether it is 
voice or unvoice is judged on the basis of the periodicity of the 
inversion of the polarity and voice-unvoice display signal is produced at 
the output terminal 6h. For this purpose, the similarity of the main 
series and the sub-polarity series to a reference polarity series of 
sampled sine and cosine waves with a fundamental and its harmonic 
frequency component is calculated by the polarity bit matrix processor 6e. 
The reference polarity series is a polarity bit sequence matrix consisting 
of elements of .+-.1 expressed by the following equations (2) and (3): 
##EQU2## 
where 
j=1.about.m=1.about.32, 
i=n.sub.1 .about.n.sub.2, 
.DELTA.f=125 Hz, 
m=unit block length. 
On the other hand, the main and sub-polarity series may be expressed as 
Z.sup.l (t) if the characteristics of the digital filter are specified by 
l(=1.about.5) as shown in Table 1. 
TABLE 1 
______________________________________ 
l k i Z.sub.j.sup.l (t) 
______________________________________ 
1 2.about.6 
3.about.12 
Sign [Xj + Xj - 1] 
2 7.about.14 
13.about.28 
Sign [Xj] 
3 15.about.18 
29.about.36 
Sign [Xj - Xj - 1] 
4 16.about.21 
31.about.42 
Sign [Xj - 2Xj - 1 + 2Xj - 2 - 
Xj - 3 + 0.5Xj - 4] 
5 20.about.30 
39.about.60 
Sign [Xj - Xj - 1 + 0.5Xj - 2] 
______________________________________ 
Therefore, the main polarity series is Z.sup.2 (t), the low-passed 
sub-polarity series is Z.sup.1 (t), and the high-passed (preemphasized) 
sub-polarity series is Z.sup.3 (t) to Z.sup.5 (t). The degree Y of the 
pattern matching of the main and sub-polarity series Z.sup.2 (t) with the 
reference polarity series H is expressed by the following equations (4) 
and (5), and the polarity pattern matching power P.sub.k.sup.l (t) may be 
expressed by the following equation (6). This polarity pattern matching 
power P.sub.k.sup.l (t) represents the similarity mentioned above. 
EQU Y.sup.l (t)=(Y.sub.i.sup.l (t)) (4) 
EQU Y.sup.l (t)=(1/m).multidot.H.multidot.Z.sup.l (t) (5) 
EQU P.sub.k.sup.l (t)=Y.sub.2k-1.sup.l (t).sup.2 +Y.sub.2k.sup.l (t).sup.2 (6) 
where 
m=32, 
k=r.sub.1 .about.r.sub.2. 
To sum up, the polarity bit matrix processor 6e makes a matrix operation of 
the main and sub-polarity series Z.sup.l (t) and the reference polarity 
series H to produce the similarity P.sub.k.sup.l (t). This similarity 
P.sub.k.sup.l (t) indicates how the original and filtered unit blocks t 
expressed by l are similar to 125 Hz and its higher harmonics expressed by 
k for each combination of l and k. The numerical value in the column k of 
Table 1 represents an example of the possible range of k for each 
characteristic l of the digital filter, and the numerical value in the 
column i of Table 1 represents the range of i=2k-1 and i=2k corresponding 
to the range of the column k. 
The similarity P.sub.k.sup.l (t) calculated by the polarity bit matrix 
processor 6e as described above is fed to the discriminator 6f and 
compared with the reference value P.sub.Thl. This reference value 
P.sub.Thl may be set by taking the filter characteristic l, frequency k 
and zero crossing rates zct, zcpt as parameters. The zero crossing rate 
zct is the zero crossing rate of the original unit block .tau., and the 
zero crossing rate zcpt is the zero crossing rate of the unit block .tau. 
pre-emphasized by the digital filter shown in FIG. 5, wherein a.sub.1 =1, 
a.sub.2 =-0.5, a.sub.3 =0. In FIG. 5, D indicates a delay device, and 
a.sub.1 to a.sub.3 are filter coefficients. These zero crossing rates zct 
and zcpt are counted by the zero crossing rate counter 6c for each unit 
block .tau. and successively fed to the threshold selector 6d. The 
reference values P.sub.Thl corresponding to the zero crossing rates zct 
and zcpt are applied from the threshold selector 6d to the discriminator 
6f. As described above, the setting of the respective values P.sub.Thl are 
made on the basis of the fact that each filter is set so that Z.sup.1 (t), 
Z.sup.2 (t), Z.sup.3 (t), Z.sup.4 (t) and Z.sup.5 (t) may be easily made 
similar to low, medium low, middle, medium high and high frequency bands 
respectively, as indicated by k and i in Table 1. Thus, if S.sub.Th 
.ltoreq.St, the nature of speech in the unit .tau. is determined by the 
discriminator 6f on the basis of the criterion as stated in the following 
item (a) or (b). (a) If in a unit block .tau. the condition that 
P.sub.k.sup.1 (t).gtoreq.P.sub.TH1 or P.sub.k.sup.2 .gtoreq.P.sub.Th2 is 
satisfied for at least one of K, it is determined that the unit block 
.tau. is a voice, and if not so, it is determined that the unit block 
.tau. is an unvoice. (b) if in a unit block .tau. either of the conditions 
that P.sub.k.sup.3 (t).gtoreq.P.sub.TH3, P.sub.k.sup.4 
(t).gtoreq.P.sub.Th4 and P.sub.k.sup.5 (t).gtoreq.P.sub.Th5 is satisfied 
for at least one of K or if neither of the conditions that P.sub.k.sup.l 
(t).gtoreq.P.sub.Th1, P.sub.k.sup.2 (t).gtoreq.P.sub.Th2, P.sub.k.sup.3 
(t).gtoreq.P.sub.TH3, P.sub.k.sup.4 (t).gtoreq.P.sub.Th4 and P.sub.k.sup.5 
(t).gtoreq.P.sub.Th5 is satisfied, it is determined that the unit block 
.tau. is an unvoice, and if not so, it is determined that the unit block 
.tau. is a voice. 
The assignment controller 10 detects the occurrence of freeze-out. If 
freeze-out occurs, predictive coding of the unvoiced unit block is 
effected with a basic bit length (i.e. 4 bits), while predictive coding of 
the voiced unit block is successively effected with a shorter bit length 
(i.e. 3 bits) than the basic bit length. Thus even if freeze-out would 
occur, a new channel can be insured by gathering the reduced bits, thereby 
avoiding the freeze-out. Since the reduction of bit is made for the unit 
block having a good signal-to quantization noise ratio (S/N.sub.q), this 
has almost no adverse effect on the quality of the whole speech. The 
relationships between DSI gain and freeze-out will be described. The 
relationships between DSI gain and the rate of occurrence of freeze-out at 
an average operating rate of 35 to 38% are that DSI gain of 3 calculated 
in terms of the basic bit length (4 bits) leads to about 10% shortage of 
channel for 240 trunks, while DSI gain given by the inverse number of an 
average operating rate of 35 to 38% leads to about 5% shortage of channel 
for 240 trunks. If the reduction to 3 bits is effected, freeze-out is 
supressed to less than about 5.5.times.10.sup.-2 % for an average 
operating rate of 38% and to less than about 3.times.10.sup.-3 % for an 
average operating rate of 35%. Therefore, freeze-out can be almost 
completely absorbed even if DSI gain of 3 is set. 
FIG. 6 shows a digital speech transmission system constructed according to 
this invention. An analog signal S.sub.in applied to the input terminal of 
the transmission side TX is passed through a low-pass filter 15 of 4 KHz, 
sampled at 8 KHz by a sampler 16, compressed by a compressor 17 and then 
quantized to 8 bit by an encoder 18, thereby being transmitted to a 
domestic digital network as a non-linear PCM signal NL of 8 KHz.multidot.8 
bit. This non-linear PCM signal NL includes a speech signal of the band of 
0.3 to 3.4 KHz, as described above. In case of transmitting this 
non-linear PCM signal NL over a long distance, the signal is converted 
into a linear PCM signal L.sub.1 of 13 bits by the expander 1, shifted 
down on its frequency band from 0.3 to 3.4 KHz to 0.1 to 3.2 KHz, passed 
through the filter 3 to leave only frequency components of 0.1 to 3.0 KHz 
and then processed by the sampling speed converter 4 to obtain a linear 
PCM signal L.sub.4 of 13 bits with 6 KHz sampling. To this linear PCM 
signal L.sub.4 of 6 KHz, predictive coding of 4 bit/3 bit and speech 
interpolation are effected for each DSI output channel by DSI device 14 
including the variable quantizing bit type ADPCM encoder 7, and the linear 
PCM signal L.sub.4 is transmitted to a digital undersea cable or the like. 
On the other hand, in the receiving side RX, digital signal of 6 KHz 
sampling in which predictive coding and speech interpolation have been 
effected with 4 bit/3 bit is converted to linear PCM signal L.sub.4 ' of 
13 bit with 6 KHz sampling by DSI device 19 including a variable 
quantizing bit type ADPCM decoder. This linear PCM signal L.sub.4 ' is the 
same as the linear PCM signal L.sub.4 from the sampling speed converter 4 
at the transmission side TX, and is recompiled by sampling unit block in 
the same trunk from each DSI channel. Therefore, the frequency components 
of the linear PCM signal L.sub.4 ' are within 0.1 to 3.0 KHz. The linear 
PCM signal L.sub.4 ' is speed-converted from 6 KHz sampling to 8 KHz 
sampling by the sampling speed converter 20. After the frequency component 
is shifted to only 200 Hz higher frequency range by the frequency 
converter 21, the linear PCM signal L.sub.4 ' is converted into a 
non-linear PCM signal NL' of 8 bits by the compressor 22. Therefore, the 
same as the linear PCM signal L.sub.3 output from the low-pass filter 3 at 
the transmission side Tx, the linear PCM signal L.sub.3 ' output from the 
sampling speed converter 20 is 8 KHz sampling.multidot.13 bits, the 
frequency component of which is 0.1 to 3.0 KHz. However, the linear PCM 
signal L.sub.1 ' is 8 KHz sampling.multidot.--bits, the frequency 
component of which is 0.3 to 3.2 KHz, and is different from the linear PCM 
signal L.sub.1 output from the expander 1 in the point that no higher 
range of frequency component of 3.2 to 3.4 KHz is included. Such 
difference in frequency component is similarly present between the 
non-linear PCM signal NL' and the non-linear PCM signal NL. After the 
non-linear PCM signal NL' is transmitted to domestic digital network, it 
is passed through the decoder 23, the expander 24 and the interpolator 25, 
whereby analog speech signal S.sub.out of 0.3 to 3.2 KHz band is produced 
at the output terminal. If the shift-down by the frequency converter 2 is 
300 Hz and the shift-up by the frequency converter 21 is 300 Hz, analog 
speech signal of 0.3 to 3.3 KHz band can be obtained. However, if the 
amount of shift is above 300 Hz, the quality of speech is outstandingly 
degraded since the lower range is turned up in the shift-down of the 
frequency converter 2. 
As described in connection with the embodiment of this invention, since the 
predictive encoder is of a variable quantizing bit type and the reduction 
of bit length in a unit block having a good signal-to-quantization noise 
ratio is made, this invention makes it possible to avoid freeze-out 
without adversely influencing the quality of speech, thereby increasing 
DSI gain to a theoretical limit. Also, the higher frequency range of 
speech is not considerably degraded even if 6 KHz sampling is used because 
of efficient utilization of transmission line. In this connection, the 
total gain to non-linear PCM signal of 8 KHz.multidot.8 bit in case of the 
basic bit length of 4 bits with 6 KHz sampling, becomes 8 since 
3.times.(64 Kb/s/24 kb/s)=8 when DSI gain is set to 3.