Browse Prior Art Database

Digital Conference Network for Companded Modulation Telephone Switching Systems

IP.com Disclosure Number: IPCOM000083003D
Original Publication Date: 1975-Mar-01
Included in the Prior Art Database: 2005-Feb-28
Document File: 3 page(s) / 81K

Publishing Venue

IBM

Related People

Assimus, AH: AUTHOR [+2]

Abstract

Conference circuits for multiparty connections in digital telephone switching systems (PABX) should utilize the digital voice information for taking this connection. In systems in which the analog voice sources (i.e., talkers) are speaking, signals from a telephone are converted to compressed or companded serial digital code.

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Digital Conference Network for Companded Modulation Telephone Switching Systems

Conference circuits for multiparty connections in digital telephone switching systems (PABX) should utilize the digital voice information for taking this connection. In systems in which the analog voice sources (i.e., talkers) are speaking, signals from a telephone are converted to compressed or companded serial digital code.

The companded modulation algorithm which created the coded bit stream, is also the algorithm used by the conference circuit for monitoring the inputs and integrating the outputs of the connected parties in a selection matrix. One such class of algorithms seen in voice systems is known as companded delta modulation.

In a companded modulation digital system, the individual bytes of voice- representative signals are not literal digital equivalents of the analog sounds and, therefore, simple addition of the companded signals representing two or more speakers would not result in a signal which truly corresponds to their combined voices. A solution to this problem is to dedicate the digital voice path to a single source at any one instant and to switch that path as other voice source(s) (talkers) are recognized.

In Fig. 1 the following are shown.

Codec 10 - The Codec converts the analog signal from a telephone to a coded digital bit stream. The code generated is dependent on the conversion algorithm. The output is transmitted serially at a 64 KHz rate

Shift Reg 12/Size Analysis 14 - The data received from Codec 10 is loaded into shift register 12. The shift register stepping and size analysis sampling is done at a 64 KHz rate. Decoders perform the step size analysis, defined by the Codec algorithm, of the voice data in the shift register 10. An up/down counter in step-size analysis 14 controlled by the decoders, contains the discrete step-size value represented by the voice data pattern. The count value is inputted to the comparator 16 to determine which telephone has the largest step size.

Comparator 16 and Matrix Control - Comparator 16 collates the size counters and determines the largest and senses if there are any equal to it.

Fig. 2 is a breakout detailing portions of Fig. 1. As shown in Fig. 2, each input to the conference network has a maximum (max) size latch 18 and prior selection latch 20 associated with it. Only one, max latch 18 and prior latch 20, can be on simultaneously...