Stereophonic voice signal transmission system

In a stereophonic voice transmission system for transmitting voice signals of a plurality of channels, a transmitting end encodes main data consisting of at least one channel voice signal among a plurality of voice signals of a plurality of channels and additional data required for reproducing the voice signals of the remaining channels from the main data, and transmits coded main data and the coded additional data to a receiving end. The receiving end decodes the coded main data, and encodes and combines the main data with the additional data to reproduce the voice signals of the remaining channels, thereby transmitting stereophonic voice signals along a transmission line of a low transmission rate, with high quality at low cost.

The present application claims priority of Japanese patent application No. 
60-191746 filed on Aug. 30, 1985 and No. 61-82840 filed on Apr. 10, 1986. 
Detailed Description of the Invention and Related Art Statement 
The present invention relates to a voice transmission system and, more 
particularly, to a stereophonic voice signal transmission system. 
Along with the development of telecommunications techniques, demand has 
recently arisen for teleconferencing systems for allowing attendants at 
remote locations to participate in a teleconference. 
Conventional teleconferencing systems are adapted to transmit and receive 
image data (e.g., television image data, electronic balckboard data, and 
facsimile data) and voice data between remote terminals, so information 
transmission cost must be desirably reduced. In particular, if data can be 
transmitted at a bit rate of 64 bps in normal subscriber lines, a 
teleconferenceing system can be realized at low cost as compared with a 
high-quality teleconferencing system using optical fibers. A solution for 
low-cost teleconferencing is deemed to be the key to popularity and 
widespread applications of teleconferencing system in small and medium 
business corporations and at home when an ISDN (Integral Service Digital 
Network) for digitizing communications systems for individual subscribers 
is established. 
In a teleconferencing system using, for example, a 64-bps transmission 
line, it is necessary to compress a large number of pieces of image and 
voice information so as not to interefere with conference proceedings. 
FIG. 13 shows an overall system configuration of a conventional 
teleconferencing system. This system comprises a microphone 1, a 
loudspeaker 2, a television camera 3 as a man-machine image interface, a 
television monitor 4, an electronic blackboard 5, a facsimile system 6, a 
telewriting device 7, a voice unit 8 for coding voice data to 16-kbps data 
or decoding 16-kbps data to voice data, a control unit 9 (to be described 
in detail), a control pad 10 for inputting instructions to the control 
unit 9, an image unit 11 for coding an image data designated by the 
control unit 9 to 48-kbps data or decoding 48-kbps data to image data, and 
a transmission unit 13 for transmitting and receiving voice and image 
signals through a 64-kbps transmission line 12. 
In a conventional teleconferencing system using a transmission line having 
a low bit rate, even monaural voice data must be compressed to 16-kbps 
data or the like by a voice data compression scheme such as an Adaptive 
Pulse Coded Modulation (ADPCM). Therefore, stereophonic voice data is not 
used in the conventional teleconferencing system. 
Stereophonic voice is desirably adapted for a conventional teleconferencing 
system to create a feeling of being a participant in a conference and to 
discriminate between speakers. 
If stereophonic voice having the above advantages is used in a conventional 
teleconferencing system using the transmission line of a low-bit rate as 
described above, a stereophonic voice transmission system shown in FIG. 14 
is required. In the stereophonic voice transmission system, right- and 
left-channel voice signals are required to double the number of 
transmission data as compared with that in the monaural mode. 
For this reason, if stereophonic voice is used in a conventional 
teleconferencing system using a low-bit rate, e.g., 64 kbps according to a 
conventional scheme, the following techniques are required: 
(a) a technique for compressing one-channel voice transmission data to 
8-kbps data; and 
(b) a technique for reducing the bit rate of image transmission data from 
48 kbps to 32 kbps. 
The technique (a) degrades voice quality, and the technique (b) results in 
voice quality degradation and/or poor service. 
According to the conventional stereophonic transmission systems, it is very 
difficult to use stereophonic voice in teleconferencing system using a 
transmission line of a low-bit rate. 
OBJECTS AND SUMMARY OF THE INVENTION 
It is therefore, a first object of the present invention to provide a 
stereophonic voice transmission system capable of high-quality 
transmission and high-quality voice reproduction in a transmission line of 
a low-bit rate rate. 
It is a second object of the present invention to provide a stereophonic 
voice transmission system comprising a small-capacity storage means and 
capable of transmitting and reproducing stereophonic voice of higher 
quality. 
It is a third object of the present invention to provide a stereophonic 
voice transmission storage system capable of transmitting and reproducing 
high-quality stereophonic voice at low cost. 
The above and other objects of the present invention will be apparent from 
the following description. 
In order to achieve the above objects of the present invention, there is 
provided a stereophonic voice transmission system for transmitting a voice 
signal among the voice signals of a plurality of channels, and additional 
data required for use together with the main data to reproduce the voice 
signals of remaining channels, and sends coded main data and coded 
additional data, and a receiving end decodes the voice signal of each 
channel sent as the coded main data and reproduces the voice signals of 
the remaining channels by the coded main data and the coded additional 
data. 
According to the present invention, in stereophonic voice transmission, 
only the main voice signal and difference signals or compressed difference 
signals are transmitted and then are received and combined by a receiving 
end. Therefore, data transmission can be performed by using a small number 
of data as compared with a conventional transmission storage system, and 
high-quality stereophonic transmission or storage can be achieved at low 
cost.

DETAILED DESCRIPTION OF THE INVENTION AND PREFERRED EMBODIMENTS 
The present invention will be described in detail by exemplifying 
teleconferencing systems with reference to the accompanying drawings. 
For the sake of simplicity, the transmission direction is represented by 
only one direction in the following description. A plurality of attendants 
or speakers (A1 to A4 or B1 to B4) in one of the conference rooms rarely 
speak simultaneously in a normal conference atmosphere. Even if the 
attendants in one conference room start to speak simultaneously, necessity 
for stereophonic voice transmission thereof is low as compared with the 
case wherein an individual attendant speaks and his voice data is to be 
transmitted. In the following description, a case will be exemplified 
wherein sounds from each speaker in the form of voice information in a 
conference are transmitted as stereophonic voice data. 
The principle of stereophonic voice transmission according to the present 
invention will be described below. 
Speaker's voice X(.omega.) (where .omega. is the angular frequency) in the 
form of a single utterance is input to right- and left-channel microphones 
1.sub.R and 1.sub.L, respectively. In this case, an echo component from a 
wall is neglected. If the right- and left-channel transfer functions are 
defined as G.sub.R (.omega.) and G.sub.L (.omega.), left- and right 
channel voice signals Y.sub.L (.omega.) and Y.sub.R (.omega.) are defined 
as follows: 
EQU Y.sub.L (.omega.)=G.sub.L (.omega.).X(.omega.) (1) 
EQU Y.sub.R (.omega.)=G.sub.R (.omega.).X(.omega.) (2) 
Substitution of equation (2) into equation (1) yields the following 
equation: 
EQU Y.sub.L (.omega.)=(G.sub.L (.omega.)/G.sub.R (.omega.)).Y.sub.R (.omega.) 
(3) 
EQU =G(.omega.).Y.sub.R (.omega.) (4) 
The above equation indicates that the voice signals of the right- and 
left-channels can be reproduced only if the transfer function G(.omega.) 
and one of the G.sub.L signal channels are known. 
According to the present invention, therefore, if the voice signal of one 
channel and a transfer function are transmitted (i.e., the voice signal of 
both channels need not be transmitted), the receiving end can reproduce 
the voice signals of right and left channels, thus realizing stereophonic 
voice transmission. In this case, the transfer function can be 
approximated by simple delay and attenuation if approximation precision is 
improved. The transfer function thus requires a smaller number of data as 
compared with the voice data Y.sub.L (.omega.) so as to achieve 
stereophonic voice transmission. 
FIG. 1 schematically shows a stereophonic voice transmission system 
according to a first embodiment of the present invention. 
The stereophonic voice transmission system comprises left- and 
right-channel microphones 1L and 1R, loudspeakers 2L and 2R, an estimator 
20 for estimating a transfer function G(.omega.), and a generator 21 for 
generating the transfer function G(.omega.) and the right-channel voice 
signal Y.sub.R (.omega.) to produce the left-channel voice signal Y.sub.L 
(.omega.). 
Referring to FIG. 1 speaker's voice X(.omega.) from the speaker A1 is input 
as the voice signal Y.sub.R (.omega.) at the right-channel microphone 1R 
and the voice signal Y.sub.L (.omega.) at the left-channel microphone 1L. 
The transmitting end transmits the right-channel voice signal Y.sub.R 
(.omega.) without modifications. The left-channel voice signal Y.sub.L 
(.omega.) is input together with the right-channel voice signal Y.sub.R 
(.omega.) to the estimator 20. The estimator 20 performs the following 
calculation to estimate the transfer function G(.omega.): 
EQU G(.omega.)=Y.sub.L (.omega.)/Y.sub.R (.omega.) 
The resultant transfer function G(.omega.) is transmitted. 
The receiving end simply receives and reproduces the transmitted 
right-channel voice signal Y.sub.R (.omega.). The transfer function 
G(.omega.) and the right-channel voice signal Y.sub.R (.omega.) are input 
to the mixer 21, and the mixer 21 performs the following operation: 
EQU Y.sub.L (.omega.)=G(.omega.).Y.sub.R (.omega.) 
so that the left-channel voice signal is reproduced. 
In this case, the transfer function G(.omega.) is derived from equations 
(3) and (4): 
EQU G(.omega.)=G.sub.L (.omega.)/G.sub.R (.omega.) (5) 
where G.sub.R (.omega.) and G.sub.L (.omega.) are right- and left-channel 
transfer functions determined by the acoustic characteristic of the room 
and the speakers' positions. G.sub.R (.omega.) and G.sub.L (.omega.) are 
not influenced by speaker's voice X(.omega.). 
The transfer function G(.omega.) is stationary according to equation (5) if 
the speaker is not changed to another location. The duration of most of 
the steady states are several hundred msec or longer. 
The speaker's voice X(.omega.) is not stationary, and therefore the left- 
and right-channel voice signals Y.sub.L (.omega.) and Y.sub.R (.omega.) 
are not stationary according to equations (1) and (2). If the transfer 
function G(.omega.) is not very complicated, i.e., if an indoor 
reverberation time is not long, the number of data required for the 
transfer function G(.omega.) is smaller than for the voice signal Y.sub.L 
(.omega.). Therefore, the technique of this embodiment which transmits the 
transfer function G(.omega.) is advantageous over the conventional 
stereophonic voice transmission technique which transmits the voice signal 
Y.sub.L (.omega.) itself. 
FIG. 2 is a schematic view of a stereophonic voice transmission system 
according to a second embodiment of the present invention. This system 
comprises left- and right-channel microphones 1L and 1R, loudspeakers 2L 
and 2R, an estimator 22 for estimating a transfer function G(.omega.) or 
impulse response H(k), a partial extractor 23 for extracting an 
approximated transfer function G(.omega.) or an approximated impulse 
response H(k), and a mixer 24 for producing the left-channel voice signal 
Y.sub.L (.omega.) using the approximated transfer function G(.omega.) or 
the approximated impulse response H(k). The impulse response of the 
transfer function G(.omega.) at the kth sampling timing is given as 
follows: 
EQU H(k)=(h-m(k), h-m+1(k), - - - , hn(k)).sup.T, 
for m&gt;0, and n&gt;0 
where T is the transposed matrix and h.sub.0 (k) is the center tap. 
In the same manner as in the first embodiment, the transmitting end sends 
the coded right-channel voice signal. The transfer function G(.omega.) or 
the impulse response H(k) derived from the right- and left-channel voice 
signals is estimated by the estimator 22. As shown in FIG. 3(A), the 
impule response H(k) has a waveform having a duration between -1,000 to 
+2,000 samples. For illustrative convenience, the speaker speaks for 500 
msec on the average. In this case, assuming that each sampled value is 
quantized with 8 bits, and that the quantized signals are transmitted, a 
transmission rate required for transmitting this impulse response is as 
high as 8.times.3,000.times.2=48 kbps. 
For this reason, as shown in FIG. 3(B), part (e.g., samples between -20 and 
+80) of the impulse response is extracted by the partial extractor 23 and 
is then transmitted. In this case, the transmission rate is 1.6 kbps, 
which is desirably lower than 16 kbps of the ADPCM. 
The stereophonic effects are determined by the phases and delay times of 
the voice. As shown in FIG. 3(C), only the position and magnitude of the 
main tap having the maximum magnitude among all taps are extracted and 
transmitted. In this case, the feeling of presence in the conference is 
slightly degraded. If eight bits are assigned to magnitude data and 
another eight bits are assigned to position data, the bit rate becomes 32 
bps, thereby greatly reducing the bit rate. 
At the receiving end, the transmitted right-channel voice signal is simply 
reproduced. The right-channel voice signal and the approximated transfer 
function G(.omega.) or the approximated impulse response H(k) are mixed by 
the mixer 24 to reproduce the left-channel voice signal. 
As described above, if the transfer function G(.omega.) is simple, it is 
advantageous to send the transfer function G(.omega.) in the place of the 
left-channel voice signal Y.sub.L (.omega.). However, if the transfer 
function G(.omega.) is very complicated, it is less advantageous to send 
the transfer function G(.omega.) in place of the left-channel voice signal 
Y.sub.L (.omega.). For this reason, in the second embodiment, the 
approximated transfer function G(.omega.) or the approximated impulse 
response H(k) is sent in place of the transfer function G(.omega.), 
thereby reducing the number of data to be sent. 
FIG. 4 is a blcok diagram of a transmitting end (A) and a receiving end (B) 
for performing transmission using an approximated impulse response. 
The transmitting end (A) comprises microphones 1L and 1R, amplifiers 25L 
and 25R, A/D converters 26L and 26R, an adaptive filler 27, delay circuits 
28L and 28R, a subtracter 29, an approximator 30, a threshold detector 31, 
level detectors 32a and 32b, a level ratio detector 33, an ADPCM circuit 
34, and a transmitter circuit 35. The receiving end (B) comprises a 
separator 36, an ADPCM circuit 37, D/A converters 38L and 38R, amplifiers 
39L and 39R, loudspeakers 2L and 2R, a coefficient circuit 40, and a 
filter 41. 
The left- and right-channel voice signals are input at the microphones 1L 
and 1R and amplified by the amplifiers 25L and 25R to predetermined 
levels. The amplified signals are sampled by the A/D converters 26L and 
26R at the sampling frequency of 1 kHz, thereby obtaining digital signals 
X.sub.L (k) and X.sub.R (k) at a kth sampling time. 
The right-channel voice signal X.sub.R (k) is input to the 256-tap adaptive 
filter. The left-channel voice signal X.sub.L (k) is delayed by the 
circuit 28L by d samples so as to adjust the position of the center tap of 
the adaptive filter 27. The delayed signal is input as X.sub.L (k-d) to 
the subtracter 29. The subtracter 29 subtracts the output X.sub.L (k) of 
the adaptive filter from X.sub.L (k-d) to produce an error or difference 
signal e(k). The adaptive filter uses the power of the error signal as an 
evaluation function and controls the tap coefficient so as to minimize the 
error power according to a known scheme such as identification by 
learning. 
When learning of the adaptive filter advances, the following results can be 
obtained: 
EQU X.sub.L (k-d).apprxeq.X.sub.L (k) 
therefore, the left-channel voice signal can be derived using the 
right-channel voice signal. 
The approximator 30 compresses the tap coefficient data of the adaptive 
filter 27 to the required bit rate. Various approximation schemes may be 
proposed, as shown in FIGS. 3(A) to 3(C). In this embodiment, the scheme 
in FIG. 3(B) is used. The approximator selects the main tap coefficient 
h100(k) among the adaptive filter tap coefficients h1(k) to h256(k) and 
quantizes 23 tap coefficients from h89(k) to h111(k) into 8-bit codes at 
the bit rate of 1 kbps. The 8-bit main tap code and an 8-bit header 
representing the start of the data string are formatted into a frame in 
FIG. 5, and the resultant frame data is sent as additional data to the 
transmitter circuit 35. 
Tap coefficient approximation in the adaptive filter 27 is performed 
whenever the speaker is changed. The threshold detector 31 detects that 
the error signal e(k) exceeds a threshold value and then becomes lower 
than the threshold value, so that a change in speaker is detected. 
This detection will be described in more detail with reference to FIG. 6. 
Referring to FIG. 6., the speaker is changed from the speaker A to the 
speaker B after the lapse of 400 msec. In this case, the transfer 
function is changed from G.sub.A (.omega.) to G.sub.B (.omega.), and the 
impulse response is changed from H.sub.A (k) to H.sub.B (k). The adaptive 
filter 27 learns to follow the change in impulse response, and the tap 
coefficient is changed. Therefore, X.sub.L (k-d).noteq.X.sub.L (k) is 
temporarily established. The level of the error single e(k) is increased. 
Thereafter, the level of the error signal e(k) is decreased below the 
threshold value, and the additional data is updated after the lapse of 600 
msec as shown in FIG. 6. 
In the transmitting end (A) in FIG. 5., the learning time of the adaptive 
filter 27 is set to be 200 msec or less. The additional data is updated 
200 msec after the actual change in speakers. For this reason, the main 
data voice signal X.sub.R (k) is delayed by the delay circuit 29R by 200 
msec so as to be synchronized with the additional data. 
Since the additional data includes some of the adaptive filter tap 
coefficients, direct transmission thereof lowers the level of the 
resultant left-channel voice signal X.sub.L (k) at the receiving end. In 
order to prevent this, the level detectors 32a and 32b and the level ratio 
detector 33 cooperate to detect a level ratio of X.sub.R (k) to X.sub.L 
(k), and the approximator 30 corrects the approximated tap coefficient, 
thereby optimizing the level of the resultant left-channel voice signal 
X.sub.L (k). Thereafter, the additional data is sent as a 1-kbps frame to 
the transmitter circuit 35. 
The transmitter circuit 35 mixes the additional data and main data obtained 
by converting the right-channel voice signal from the delay circuit 28R 
into a 15-kpbs ADPCM code from the ADPCM circuit 34 to produce a time 
compressed 16-kbps stereophonic voice data. This voice data is transmitted 
to the receiving end (B) through a transmission line. 
The stereophonic voice data sent to the receiving end (B) is separated by 
the separator 36 into the 15-kbps right-channel voice signal of the main 
data and the 1-kbps additional data. The right-channel voice signal is 
decoded by the ADPCM circuit 37, and the decoded signal is converted by 
the D/A converter 38R to an analog signal. The analog signal is then 
amplified by the amplifier 39R and produced at the loudspeaker 2R. 
On the other hand, the additional data is converted by the coefficient 
circuit 40 into 256 tap coefficients. These coefficients are supplied to 
the filter 41. The filter 41 uses the tap coefficients and the 
right-channel voice signal to produce the left-channel voice signal. The 
resultant left-channel voice signal is converted by the D/A converter 38L, 
amplified by the amplifier 39L, and produced at the loudspeaker 2L in the 
same manner as the right-channel voice signal. 
Thus, the transmitted time-compressed data is time-expanded by the 
receiving end. 
The above embodiment can be realized by using the current techniques, and 
the number of stereophonic signals necessary to be trnsmitted can be 
greatly reduced. 
In this embodment, the adaptive filter 27 and the filter 41 comprises 
transversal filtersof time region processing. However, these filters may 
be replaced with filters of frequency region processing to achieve the 
same effect as described above. 
In addition, a correlator may be used in place of the adaptive filter to 
detect a tap having a maximum correlation value. 
FIG. 7 schematically shows a configuration of a stereophonic voice 
transmission system according to a third embodiment of the present 
invention. The stereophonic voice transmission system comprises left- and 
right-channel microphones 1L and 1R, loudspeakers 2L and 2R, an estimator 
22 for estimating a transfer function G(.omega.) or an impulse response 
H(k), a partial extractor 23 for extracting an approximated transfer 
function G(.omega.) or an approximated impulse response H(k), tables 42a 
and 42b for prestoring reference approximated transfer functions 
G(.omega.) and reference approximated impulse responses H(k), an encoder 
43, a decryptor 44, and a generator 24 using the right-channel voice 
signal and the approximated transfer function G(.omega.) or the 
approximated impulse response H(k) to produce the left-channel voice 
signal. 
In this embodiment, in the transmitting end, the reference approximated 
transfer function F(.omega.) or the reference approximated impulse 
response H(k) from the table 42a is compared by the encoder 43 with the 
approximated transfer function G(.omega.) or the approximated impulse 
response H(k) extracted from th partial extractor 23. A code g 
representing the highest similarity between the prestored and the 
extracted data is transmitted. At the receiving end, the decryptor 44 
receives the code g and the data read out from the table 42b to produce 
the approximated transfer function G(.omega.) or the approximated impulse 
response H(k). The output from the decryptor 44 and the right-channel 
voice signal are mixed by the mixer 24 to produce the left-channel voice 
signal. According to this embodiment, if the acoustic characteristics in 
the conference room are known, the number of transmission signals are 
reduced, while the feeling of presence in a conference is maintained. 
FIG. 8 schematically shows a stereophonic voice transmission system 
according to a fourth embodiment of the present invention. 
A transmitting end in this stereophonic voice transmission system comprises 
left- and right-channel microphones 101R and 101L, a delay circuit 120 for 
delaying the left-channel microphone input voice signal, an estimator 121 
for producing an estimated left-channel voice signal y(k) from the 
right-channel voice signal x(k) in the time region, and a subtracter 122 
for subtracting the estimated left-channel voice signal y(k) from the 
left-channel voice signal y(k). A receiving end comprises left- and 
right-channel loudspeakers 102L and 102R, a mixer 123 for producing an 
estimated left-channel voice signal y(k) from the right-channel voice 
signal x(k), and an adder 124 for adding a difference signal to the 
left-channel voice signal y(k) estimated by themixer 123. The transmitting 
and receiving ends are connected through transmission lines 125 and 126. 
It should be noted x(k) and y(k) show values of the left- and 
right-channel voice signals at the kth sampling time. 
Referring to FIG. 8, voice X(.omega.) output by a speaker A1 is input to 
the microphones 101R and 101L, and microphone input signals Y.sub.R 
(.omega.) and Y.sub.L (.omega.) are represented by transfer functions 
F.sub.R (.omega.) and G.sub.L (.omega.) determined by the propagation 
delays and the acoustic characteristics of the room. In this case, .omega. 
is the angular frequency. 
EQU Y.sub.R (.omega.)=F.sub.R (.omega.).X(.omega.) (101) 
EQU Y.sub.L (.omega.)=G.sub.L (.omega.).X(.omega.) (102) 
The left microphone input signal Y.sub.L (.omega.) is delayed with 
C(.omega.) in the delay circuit 120 so as to guarantee the 
cause-and-effect relationship in the estimator 121 and is represented by 
the transfer function F.sub.L (.omega.) for an arrangement including 
components up to the delay circuit 120: 
EQU Y.sub.L (.omega.)=C(.omega.).G.sub.L (.omega.).X(.omega.)=F.sub.L 
(.omega.).X(.omega.) (103) 
The left-channel voice signal Y.sub.L (.omega.) is input to the subtracter 
122. 
The estimator 121 uses the left- and right-channel voice signals Y.sub.R 
(.omega.) and Y.sub.L (.omega.) is estimate the transfer function 
G(.omega.) for deriving the left-channel voice signal Y.sub.L (.omega.) 
from the right-channel voice signal Y.sub.R (.omega.) as follows: 
EQU G(.omega.)=F.sub.L (.omega.)/F.sub.R (.omega.) (104) 
so that the estimated transfer function G(.omega.) can be obtained. 
The estimator 121 mainly includes an adaptive transversal filter 121a for 
calculating the estimated left-channel voice signal y(k) in the time 
region of FIG. 9(A), and a correction circuit 121b for sequentially 
updating the estimated input response H(k) of the transfer function 
G(.omega.) shown in FIG. 9(B). The adaptive transversal filter 121a and 
the correction circuit 121b are operated in synchronism with clocks. 
The adaptive transversal filter 121a comprises n tap shift registers 127, 
multipliers 128 for multiplying the components of the estimated impulses 
response H(k) with the corresponding components of the right-channel voice 
signal X(k), and an adder 129 for adding outputs of the multipliers 128. 
The components of the right-channel voice signal input to the shift 
registers each having a delay time corresponding to one sampling time so 
that time-serial vector X(k) is produced as follows: 
EQU X(k)=(X(k), X(k-1), . . . , X(k-N+1)).sup.T (105) 
where T is the transposed vector. 
If the estimated impulse response obtained by approximating the estimated 
transfer function G(.omega.) in the time region is given as follows: 
EQU H(k)=(h1(k), h2(k), h3(k), . . . , hN(k)).sup.T (106) 
an estimated value y(k) of the left-channel voice signal y(k) can be 
obtained below: 
EQU y(k)=H(k).sup.T.X(k) (107) 
In this case, if the impulse response series H of the transfer function 
G(.omega.) is expressed as: 
EQU H=(h1, h2, . . . , hN).sup.T (108) 
and the transfer function can be effectively estimated as: 
EQU H(k)=H (109) 
then the left-channel voice signal estimated value y(k) is an approximated 
of the actual left-channel voice signal y(k). 
Estimation of the impulse response H(k) in the estimator 121 is performed 
by causing the correction circuit 121b to sequentially performed the 
following operation: 
EQU H(k+1)=H(k)+.alpha.e(k).X(k)/.parallel.X(k).parallel..sup.2 (110) 
for H(0)=0 
The above algorithm is a known identification technique by learning. In 
equation (110), e(k) is the output from the subtracter 122 and given as 
follows: 
EQU e(k)=y(k)-y(k) (111) 
and is the coefficient for determining a convergence rate and stability. 
As a result, only the difference signal e(k) is sent as the left-channel 
data at the end of the above operations. 
The receiving end has the generator 123 having the same arrangement as that 
of the estimator 121. The generator 123 sequentially traces the estimation 
results of the left-channel voice signals from the transmitting end 
according to the right-channel voice signal X(k) and the difference signal 
e(k) and calculates the estimated left-channel voice signal y(k) (Y.sub.L 
(.omega.) in the frequency region basis) from the following equation: 
EQU y(k)=H(k).sup.T.X(k) (112) 
EQU H(k+1)=H(k)+.alpha.e(k).X(k)/.parallel.X(k).parallel..sup.2 (113) 
where H(k) is the estimated tap coefficient series in the generator and 
H(0)=0. 
Equations (112) and (113) in the receiving end are the same as equations 
(107) and (110) in the transmitting end, so that the estimated values y(k) 
and y(k) of the transmission and reception of the left-channel voice 
signal are given as follows: 
EQU y(k)=y(k) (114) 
The left-channel output y.sub.L in the receiving end, therefore, is given 
as a sum of the estimated value y(k) and the difference signal e(k) from 
the adder 124: 
EQU y.sub.L (k)=y(k)+e(k)=y(k)+e(k)=y(k) (115) 
As a result, the left-channel voice can be properly reproduced. 
The estimator 121 and the generator 123 in the stereophonic voice 
transmission system are adaptive transversal filters of the time region. 
However, these filters may be replaced with adaptive filters of the 
frequency region to obtain the same result as described above. 
According to this scheme, the accurate more the approximation of the 
estimated values y(k) and y(k) of the left-channel voice, the less the 
power of e(k) becomes. The number of bits of e(k) can be smaller than that 
of y(k). 
A fifth embodiment of the present invention will be described below. 
FIG. 10 schematically shows the configuration of a stereophonic voice 
transmission system of this embodiment. 
In this embodiment, after a correlation component between the right- and 
left-channel voice signals is removed, the data is coded and decoded 
according to the ADPCM scheme. Transmission and storage of stereophonic 
voice can be performed using a small number of data. The same reference 
numerals as in FIG. 8 (the fourth embodiment) denote the same parts in 
FIG. 10. 
Referring to FIG. 10, a transmitting end comprises a right-channel ADPCM 
encoder unit 130 and a left-channel ADPCM encoder unit 131. A receiving 
end comprises a right-channel ADPCM decoder unit 132 and a left-channel 
ADPCM decoder unit 133. 
An interchannel correlation eliminator unit 134 has the substantially same 
function as that of the estimator 121 in FIG. 8, and an interchannel 
correlation adder unit 135 has the substantially same function as the 
mixer 124 in FIG. 8. 
Single utterance voice S(.omega.) (where .omega. is the angular frequency) 
is input at right- and left-channel microphones 1R and 1L with right- and 
left-channel transfer functions F.sub.R (.omega.) and F.sub.L (.omega.) 
determined by the acoustic characteristics of the room and is converted 
toright- and left-channel voice signals X(.omega.) and Y(.omega.). 
The interchannel correlation eliminator unit 134 in the transmitting end 
causes the ADPCM encoder units 130 and 131 (to be described in detail 
later) to produce the estimated transfer function G(.omega.) according to 
an encoded right-channel voice signal X.sub.1 (.omega.) and a difference 
signal E.sub.1 (.omega.) as follows: 
EQU G(.omega.)=Y(.omega.)/X(.omega.)=F.sub.L (.omega.)S(.omega.)/F.sub.R 
(.omega.)S(.omega.)=R.sub.L (.omega.)/F.sub.R (.omega.) (116) 
A predicted value Y(.omega.) is derived from: 
EQU Y(.omega.)=G(.omega.)X.sub.1 (.omega.) (117) 
and is subtracted by the subtracter 122 from the left-channel voice signal 
Y(.omega.) to produce a predicted difference signal E(.omega.). The signal 
E(.omega.) is then input to the left-channel ADPCM encoder unit 131. 
The encoder unit 131 sends the ADPCM coded difference signal F(.omega.) 
onto a transmission line 126. The encoder unit 131 has the same decoding 
function as the ADPCM encoder unit 133. A decoded difference signal 
E.sub.1 (.omega.) of the difference signal F(.omega.) decoded by the 
decoding function of the encoder unit 131 is input to the correlation 
eliminator unit 134. 
The right-channel ADPCM encoder unit 130 sends the ADPCM encoded 
right-channel voice signal D(.omega.) onto a transmission line 125. The 
right-channel ADPCM encoder unit 130 has the same decoding function as 
that of the ADPCM decoder unit 132. A decoded signal X.sub.1 (.omega.) of 
the right-channel voice signal D(.omega.) decoded by the decoding function 
of the encoder unit 130 is input to the correlation eliminator unit 134. 
At the receiving end, D(.omega.) and F(.omega.) are ADPCM decoded by the 
ADPCM decoder units 132 and 133, and the interchannel 135 performs the 
following calculations: 
EQU Y(.omega.)=G(.omega.)X.sub.1 (.omega.) (118) 
EQU Y.sub.L (.omega.)=E.sub.1 (.omega.)+Y(.omega.) (121) 
therefore the left-channel voice signal Y.sub.L (.omega.) is reproduced. In 
other words, the interchannel correlation adder unit 135 receives the 
decoded X.sub.1 (.omega.) and E.sub.1 and estimates G(.omega.). 
FIG. 11 is a detailed circuit diagram of this embodiment. 
A transmitter end 200 in the stereophonic voice transmission system 
comprises a right-channel ADPCM encoder unit 130 for ADPCM encoding 
right-channel voice input at a right-channel microphone 101R, a 
left-channel ADPCM encoder unit 131 for ADPCM encoding left-channel voice 
y(t) input at a left-channel microphone 101L, and an interchannel 
correlation eliminator unit 134 for predicting the left-channel voice 
signal from the right-channel voice signal and eliminating the 
interchannel correlation component from the left-channel voice signal. 
The ADPCM signals D(k) and F(k) output from the transmitter end 200 are 
sent to a receiving end 300 through transmission lines 125 and 126. 
The receiving end 300 in the stereophonic voice transmission system 
comprises a right-channel ADPCM decoding unit 132 for decoding the 
right-channel ADPCM code D(k) and reproducing the right-channel voice 
signal X.sub.1 (t), a left-channel ADPCM decoder unit 133 for decoding the 
left-channel ADPCM code F(k) and reproducing the left-channel correlation 
adder unit 135 for predicting the left-channel voice signal from the 
right-channel voice signal and adding the interchannel correlation 
component to the ADPCM decoded voice signal. 
The respective components are described in detail below. 
Right-Channel ADPCM Encoder Unit 130: 
The right-channel ADPCM encoder unit 130 comprises an A/D converter 201, a 
subtracter 203, an adaptive quantizer 205, an adaptive dequantizer 207, an 
adder 209, and an estimator 211. 
The digitized right-channel voice signal x(k) is subtracted by the 
estimator 211 from the predicted right-channel predicted voice signal x(k) 
to obtain a predicted difference signal d(k) representing a decrease in 
power lower than that of x(k) due to prediction. The difference signal 
d(k) is encoded by the adaptive quantizer 205 to an ADPCM code having a 
bit rate of about 32 kbps so as to adaptively vary the quantization step 
according to the amplitude of the input signal. 
The estimator 211 causes the adder 209 to add the reproduced predicted 
difference signal d.sub.1 (k) decoded by the adaptive dequantizer 207 and 
the output x(k) from the estimator 211 to input the same right-channel 
reproduced voice signal X.sub.1 (k) as in the right-channel ADPCM encoder 
unit 132 in the receiving end. Adaptive filtering is performed to minimize 
the power of the reproduced predicted difference signal d.sub.1 (k). 
Left-Channel ADPCM Encoder Unit 131: 
The left-channel ADPCM encoder unit 131 comprises an A/D converter 213, a 
delay circuit 215, a subtracter 217, an adaptive quantizer 319, an 
adaptive dequantizer 321, an adder 223, and an estimator 227. 
The interchannel correlation eliminated signal e(k) obtained by eliminating 
the correlation component in the interchannel correlation eliminator unit 
134 from the left-channel voice signal delayed by the delay circuit 215 is 
ADPCM encoded in the same manner as the right-channel voice signal. 
The above-mentioned delay operation guarantees the cause-and-effect 
relationship of the right- and left-channel voice signals X(k) and y(k) 
(even if voice reaches the left-channel microphone faster than the voice 
arrival to the right-channel microphone, the left-channel voice signal is 
delayed for the input to the interchannel correlation unit 134). 
The self-correlated voice component is eliminated by the estimator 227, and 
the correlated component of the right-channel voice signal mixed with the 
left-channel voice signal is eliminated by the interchannel correlation 
eliminator unit 134. Therefore, the left-channel voice signal can be 
compressed to an ADPCM code (e.g., above 16 kbps) shorter than the bit 
length of the right-channel voice data. 
Interchannel Correlation Eliminator Unit 134: 
The interchannel correlation eliminator unit 134 comprises an estimator 229 
and a subtracter 231. The estimator 229 receives the right-channel 
reproduced voice signal X.sub.1 (k) and produces the interchannel 
correlation component y(k). The estimator 229 performs adaptive filtering 
to optimize the characteristics of the filter characteristics so as to 
minimize the power of the left-channel predicted difference signal e.sub.1 
(k) in the receiving end 300. 
Right-Channel ADPCM Decoder Unit 132: 
The right-channel ADPCM decoder unit 132 comprises an adaptive dequantizer 
233, an adder 235, an estimator 237, a delay circuit 239, and a D/A 
converter 241. 
The received ADPCM code D(k) is converted into the right-channel reproduced 
predicted difference signal d.sub.1 by the adaptive dequantizer 233. The 
signal d.sub.1 is added by the adder 235 to the right-channel predicted 
signal X(k) output from the estimator 237, thereby producing the 
right-channel reproduced voice signal X.sub.1 (k). 
Thereafter, in order for the encoder 200 to compensate for the left-channel 
delay effected by the delay circuit 239, the same length of delay time is 
added to the right-channel voice signal, is converted by the D/A converter 
241, and is output at the loudspeaker 102R. 
The estimator 237 receives the right-channel reproduced voice signal 
X.sub.1 and performs adaptive filtering for producing a minimized 
predicted right-channel difference signal d.sub.1 (k). 
The estimator 237 is the same as the estimator 211 in the right-channel 
ADPCM encoding unit 130 and receives the same signal as therein. 
Therefore, the transmitting and receiving ends 200 and 300 output the same 
predicted signals X(k). 
Left-Channel ADPCM Decoder Unit 133: 
The left-channel ADPCM decoder unit 133 comprises an adaptive dequantizer 
243, an adder 245, an estimator 247, and a D/A converter 249. 
In the same manner as in the right-channel operation, the left-channel 
predicted difference signal e.sub.1 (k) at the receiving end is produced 
from the received ADPCM code F(k). 
Thereafter, this signal is added by the interchannel correlation adder unit 
135 to the correlation component y(k) to obtain the left-channel 
reproduced voice signal y.sub.1 (k). This signal is converted by the D/A 
converter 249 into an analog signal. The analog signal is output at the 
loudspeaker 102L. 
Interchannel Correlation Adder Unit 135: 
The interchannel correlation adder unit 135 comprises an estimator 251 and 
an adder 253. The adder unit 135 receives the right-channel reproduced 
voice signal X(k) and causes the estimator 251 to produce the interchannel 
correlation component y(k). 
The estimator 251 has the same arrangement as in the transmitted end 200. 
The estimator 251 comprises an adaptive filter for learning to minimize 
the power of the receiving-end left-channel predicted difference signal 
e.sub.1 in the same manner as in the transmitting end 200, thereby 
obtaining the same predicted value y(k) as in the transmitting end 200. 
The estimator 229 (211, 227, 237, 247, or 251), the adaptive quantizer 205 
(219), and the adaptive dequantizer 233 (207, 221, or 243) will be 
described in more detail. 
Estimator 229: 
Extensive studies have made on the types and arrangements of the estimator 
229. For example, the estimator can perform prediction in the time region 
or the frequency region (e.g., FFT or Fast Fourier Transform). According 
to the present invention, any adaptive filter may be employed as the 
estimator 229. However, an adaptive transversal filter of the time region 
in FIG. 8 is used to constitute the interchannel estimator 229. 
In the following description, X.sub.R (.omega.) and e(k) are substituted by 
X.sub.1 (.omega.) and e.sub.1 (k), respectively. 
The estimator 229 mainly includes the adaptive transversal filter 121a for 
calculating the estimated left-channel voice signal y(k) in the time 
region of FIG. 8A and the correlation circuit 121b for sequentially 
correcting the estimated impulse responses H(k) of the interchannel 
transfer function G(.omega.). The adaptive transversal filter 121a and the 
correction circuit 121b are operated in synchronism with the sampling 
clocks. 
The adaptive transversal filter 121a comprises n tap shift registers 127, n 
multiplifers 128 for multiplying the components of the estimated impulse 
response H(k) with the corresponding components of the right-channel voice 
signal X(k), and an adder 129 for adding the outputs from the multipliers 
128. 
In the estimator 229, the respective components of the right-channel voice 
signal X.sub.1 are input to the shift registers 127 each having a 
one-sampling delay time so that the time serial vector is produced as 
follows: 
EQU X.sub.1 (k)=(x.sub.1 (k), x.sub.1 (k-1), . . . , x.sub.1 (k-N+1)).sup.T ( 
120) 
where T is the transposed vector. 
On the other hand, if the estimated impulse response obtained by 
approximating the estimated transfer function G(.omega.) in the time 
region is defined as follows: 
EQU H(k)=(h1(k), h2(k), h3(k), . . . hN(k)).sup.T (121) 
the estimated value y(k) of the left-channel voice signal y(k) is given as 
follows: 
EQU y(k)=H(k).sup.T.X.sub.1 (k) (122) 
In this case, if the impulse response series of the transfer function 
G(.omega.) is 
EQU H=(h1, h2, . . . , hN).sup.T (123) 
and the transfer function can be effectively estimated as 
EQU H(k).apprxeq.H (124) 
then the left-channel voice signal estimated value y(k) is a good 
approximate of the actual left-channel voice signal y(k). 
Estimation of the impulse response H(k) in the estimator 229 is performed 
by correction circuit 121b according to the following calculation for 
sequentially minimizing the power of e.sub.1 (k): 
EQU H(k+1)=H(k)+.alpha.e.sub.1 (k).multidot.X.sub.1 (k)/.parallel.X.sub.1 
(k).parallel..sup.2 (125) 
or H(0)=0 
This algorithm is known as the learning Identification Method. 
In equation (125), e.sub.1 (k) is a reproduced signal at the receiving end 
of an output (equation (126)) from the subtracter in FIG. 11: 
EQU e(k)=y(k)-y(k) (126) 
and the coefficient for determining the convergence rate and stability in 
equation (125). 
Adaptive Quantizer 205 and Adaptive Dequantizer 233: 
FIG. 12 shows the configuration of the adaptive quantizer 205 and the 
adaptive dequantizer 233. 
The adaptive quantizer 205 comprises a divider 255, an encoder 257, a 
decoder 259, a multiplier 261, and a power detector 263. The adaptive 
dequantizer 207 comprises a decoder 265, a multiplier 267, and a power 
detector 269. 
For example, a 14-bit linear predicted difference signal d(k) in the 
transmitting end is divided by a quantization step .DELTA.(k) and 
quantized. The quantized signal is encoded by the encoder 257 to an ADPCM 
code D(k) which is then sent onto the transmission line 125. 
The signal decoded by the decoder 259 is multiplied by the multiplier 261 
with the quantized step .DELTA.(k) to produce a dequantized signal d.sub.1 
(k). The power detector 263 detects the power of the signal d.sub.1 (k). 
By detecting this power, the quantization step .DELTA.(k) is determined. 
On the other hand, in the adaptive dequantizer 233, the ADPCM code is 
decoded by the decoder 265, and the decoded signal is multiplied by the 
multiplier 267 with the quantization step .DELTA.(k), thereby producing 
the 14-bit receiving-end linear predicted difference signal. The 
quantization step .DELTA.(k) is determined by detecting the power of 
d.sub.1 (k) in the power detector 269 in the same manner as described 
above. 
The above operations can be performed in the logarithmic region. 
According to the fifth embodiment, the main voice signal and the difference 
signal are ADPCM encoded and the encoded signals are transmitted in 
stereophonic transmission. As compared with the stereophonic voice 
transmission, stereophonic transmission can be achieved by a smaller 
number of signals. 
It is also possible to convert Adaptive Predictive Coding (APC) signals 
into stereophonic signals in the same manner as described above. In this 
case, the eliminated correlation coefficient in the ADPCM scheme is also 
sent to the receiving end. 
In the fourth and fifth embodiments, the estimation algorithm is 
exemplified by identification by learning. However, a steepest descent 
method may be used in place of identification by leaning. 
In the above embodiments, two-channel stereophonic voice transmission is 
exemplified. However, the present invention is not limited to such 
transmission, but may be extended to stereophonic voice transmission of 
three or more channels and is also applicable to voice storage as well as 
voice transmission 
According to the present invention as described above, only the main data 
consisting of the voice signal and the additional data required for 
reproducing the voice signals of the remaining channels in cooperation 
with the main data are sent to the receiving end. The number of data 
signals is not greatly increased as compared with the case of the monaural 
voice transmission. Therefore, a low-cost high-quality stereophonic voice 
transmission can be achieved even along a transmission line of a low 
transmission rate.