Method and circuit arrangement for allowing telephone conversations among several users

A method and circuit arrangement for allowing and improving the telephone conversation among several users in digital switched telephone PCM systems, using a multiplication coefficient inversely proportional to the level of each line.

TECHNICAL FIELD 
The invention relates to a method and to a circuit arrangement for allowing 
telephone conversations among several users in a digital switched 
telephone system, in particular in Pulse Code Modulation (PCM) Systems. 
BACKGROUND OF THE INVENTION 
When more than two users have to talk to each other it is necessary to 
apply special techniques to the telecommunication equipment so that each 
receives the audio signals from all others. 
The simplest technique applicable in this circumstance is to sum all 
received signals and send this sum-signal to the conferees. However, this 
technique provides a signal with too high a background noise and does not 
allow clear listening, in particular when several users speak 
simultaneously and therefore their voices overlap. 
Another technique for improving the preceding one is known which 
determines, at first, through suitable rules, who is the user having the 
highest signal level; then the signals from the conferees are summed 
leaving the highest-level signal unaltered and the signals from the other 
users are attenuated by a prefixed value. 
This technique, even if it partially solves the above-mentioned problems, 
does not eliminate those problems due to high differences in the signal 
level between the parties, such as those due to differences in the state 
of the individual lines that could alter the choice of the user with the 
highest signal. The resulting bad working becomes more apparent as the 
number of the users, and therefore of the lines, increases. For this 
reason, the maximum number of users allowed by this technique is 
restricted to only a few units. 
In the case where PCM digital networks are considered, signals are coded in 
digital values according to a non-linear, approximately logarithmic, 
characteristic. In this circumstance it is possible to carry out the 
operations described above only when the samples are coded in a linear 
manner. Therefore it is necessary to carry out the appropriate 
conversions. 
SUMMARY OF THE INVENTION 
It is an object of the present invention to provide a technique capable of 
overcoming the drawbacks of the known art. 
This object is achieved, in accordance with the present invention, through 
a method for allowing telephone conversations among several users in a 
digital switched telephone system, said method comprising the steps of 
receiving samples of a speech signal of each one of said users, said 
samples being non-linearly encoded, converting said non-linearly encoded 
samples into linearly encoded samples, processing said linearly encoded 
samples to thereby generate processed linearly encoded samples, summing 
said processed linearly encoded samples over said several users to obtain 
a sum of processed linearly encoded samples, and converting said sum of 
processed linearly encoded samples into a non-linearly encoded sum of 
processed samples, characterized in that, to perform said processing, said 
linearly encoded samples are varied in level through multiplication 
thereof with a coefficient which is calculated for each user and which is 
inversely proportional to the level of said speech signal received from 
said user. 
It is also achieved by a circuit arrangement for allowing telephone 
conversations among several users in a digital switched telephone system, 
said circuit arrangement including reception means adapted to receive 
samples of a speech signal of each one of said users, said samples being 
non-linearly encoded, first conversion means adapted to convert said 
non-linearly encoded samples into linearly encoded samples, processing 
means for processing said linearly encoded samples and to thereby generate 
processed linearly encoded samples, summing means for summing said 
processed linearly encoded samples over said several users to obtain a sum 
of processed linearly encoded samples, and second conversion means for 
converting said sum of processed linearly encoded samples into a 
non-linearly encoded sum of processed samples, characterized in that said 
processing means comprises multiplication means for varying the level of 
said linearly encoded samples by multiplication thereof with a coefficient 
which is calculated for each one of said users, and coefficient 
calculating means for calculating for each one of said users said 
coefficient so that said coefficient is inversely proportional to the 
level of said speech signal received from said user. 
By varying the level of each signal in a different manner for each user, 
the abovementioned drawbacks are eliminated. Through this technique it is 
possible to allow the conversation of more than thirty users. All the 
lines of the users are set to the same condition, whatever the original 
signal and the state of the individual lines may be. 
All interferences and noise of the various lines are thus eliminated. Hence 
a better understanding of the speech of the various users is obtained. 
Moreover, the structure of the system implementing such technique is 
modular and therefore insertable in any telecommunication equipment and 
can further be used in combination with the known techniques described 
above.

BEST MODE FOR CARRYING OUT THE INVENTION 
In the following reference will be made to a not limiting example of a PCM 
network and therefore operations on samples of signals of each user will 
be carried out. 
With reference to the FIG., the signal samples are converted into linear 
samples through converter 11 (A- or .mu.-law to linear-law conversion), 
samples are then processed in block 12 in accordance with the invention, 
then they are summed and optionally further processed in a known manner in 
block 13, and converted again into non-linear samples in block 14 (linear 
to A- or .mu.-law conversion). Obviously the conversion law may change 
according to the reference standard. 
All these blocks are connected to each other as indicated. The conversion 
from A- or .mu.-law to linear law and vice versa is well known to those 
skilled in the art and is not described in detail here. 
In accordance with the invention, associated with each user is a 
coefficient to obtain the level variation through multiplication by the 
speech samples of the various users. 
At the beginning, all coefficients are set to 1, i.e. no gain and no 
attenuation. Associated with each user (channel) is a score, computed as a 
sum of the absolute values of N speech samples. The score is then suitably 
compared with a predetermined threshold to determine if there is speech 
activity by the user. For the circumstance when the sum value is lower 
than such a threshold, the coefficient for the level control is left 
unchanged, otherwise it is updated according the following formula: 
EQU C.sub.i =K.sub.1 C.sub.(i-1) -K.sub.2 S+K.sub.3 
where Ci is the coefficient at the i.th instant, 
C(i-l) is the coefficient computed at the preceding instant, 
S is the sum of the absolute value of N samples of said signals, and 
K1, K2, K3 are suitable constants to be suitably optimized. 
The new coefficient is updated each time a new score is obtained, i.e. 
every N frames; it depends on the computed preceding value multiplied by a 
suitable oblivion factor (K.sub.1), on a constant (K.sub.3) and on the 
score multiplied in turn by a suitable coefficient (K.sub.2). The higher 
the score (the higher the level) the lower the level coefficient and vice 
versa. 
It is to be pointed out that it is not necessary to carry out frequent 
updatings of the level coefficient since, once the quality (level) of the 
signal is determined, it will remain approximately the same during the 
whole conversation. Moreover, the use of the activity threshold avoids the 
amplification of the background noise when no speech signal is present. 
The improvements obtained during the use of the level control allow the 
application of this conference system to the entire bundle of PCM 
channels. In particular, FIG. 1 shows a circuit arrangement for allowing 
telephone conversations among several users in a digital switched 
telephone system. The circuit arrangement includes a reception means 10, a 
first conversion means 11, a processing means 12, a summing means 13, and 
a second conversion means 14. The reception means 10 receives samples of a 
speech signal of each one of the users, said samples being non-linearly 
encoded. The first conversion means 11 converts the non-linearly encoded 
samples into linearly encoded samples. The processing means 12 processes 
the linearly encoded samples and thereby generates processed linearly 
encoded samples. The summing means 13 sums the processed linearly encoded 
samples over said several users to obtain a sum of processed linearly 
encoded samples. The second conversion means 14 converts the sum of 
processed linearly encoded samples into a non-linearly encoded sum of 
processed samples. 
As shown in FIG. 2, the processing means 12 includes a multiplication means 
15 and a coefficient calculating means 16. The multiplication means 15 
varies the level of the linearly encoded samples by multiplication thereof 
with a coefficient which is calculated for each one of the users. The 
coefficient calculating means calculates for each one of the users the 
coefficient so that the coefficient is inversely proportional to the level 
of the speech signal received from the user.