WO1991015916A1

WO1991015916A1 - Automatic system for the playback of recorded spoken messages through a telephone line

Info

Publication number: WO1991015916A1
Application number: PCT/BR1991/000004
Authority: WO
Inventors: Silvio Mauro Diaz Horta; Joaquim Marques Da Fonseca; Carlos Boromeu Lopes; Sandra Miranda Resegue Lopes
Original assignee: Mentel-Mensagens Telefonicas E Empreendimentos Ltda
Priority date: 1990-04-04
Filing date: 1991-04-04
Publication date: 1991-10-17
Also published as: BR9002488A; EP0482142A1

Abstract

This system can either replace or be coupled to the tone and ring generating equipment currently in use at telephone exchanges and is comprised of the following: input generating module, output generating module, modulating circuit, demodulating circuit, input interface, output interface, control circuit, RAM memory, mixing circuit and background music circuit. It uses the time periods normally assigned to the transmission of supervisory tones as well as other time periods during a telephone call also for transmitting spoken messages and background music concomitantly or not with the supervisory tones.

Description

Descriptive Report on the Invention Patent: "AUTOMATIC SYSTEM FOR THE PLAYBACK OF RECORDED SPOKEN MESSAGES THROUGH A TELEPHONE LINE".

This Patent has been granted for an an automatic system that concomitantly plays back and/or transmits recorded spoken messages and backgroud music. This system can either replace or be coupled to the tone and ring generating equipment currently in use in public and private telephone •exchanges and offers technical, practical and functional advantages .

The system is designed for the purpose of using the time period normally taken for transmitting supervisory tones alone (ringing, busy line, dialling, unobtainable no,etc) or any other time periods during a phone call, for concomitantly transmitting recorded spoken messages and background music. It, thus, constitutes a development of the equipment described in Patent BRMU 6802019 "EQUIPMENT FOR THE PLAYBACK OF SPOKEN MESSAGES THROUGH A TELEPHONE LINE". The utilization of this system will allow the transmission of, for example, educational, promotional or corporate messages, with or withou repetition, during any time period, 24 hours per day,through several channels. In other that the principle governing this system be understood, herebelow is a description of the typical equipment unit deemed ideal for the kind of service in question. The unit is also shown in the following attached drawings: Figure 1: Equipmet Block Diagram Figuer 2: Modulating Circuit Figure 3: Demodulating Circuit Figure 4: Input Interface Figure 5: Output Interface Figure 6: Control Unit Figure 7: RAM Memory

Figure 8: Mixing Circuit

Figure 9: Alternative Equipment Block Diagram Figure 1 shows the block diagramof the equipment comprised of an input control module, output controlmodule, memory interfaces, mixing circuit and background musical circuit. The basic functions of the input control module are: reception and transmission of recorded messages to the mass memory, filing of messages, transfer of messages to the output control module at prederminated times, system testing and control.

Once the messages have been recorded, they are codified in pulse tranis (binary signals) during amessage playback by the delta modulation circuit. The codified messages are then transferred to the mass memory, through the input interface that controls the rate of information transfer between the circuit modulator and the inputcontrolmodule. The size of the mass memory is directly proportional to the summation of the time periods of the recordedmessages. The basic functions of the output control module are: communication with the input control module for the transferof messages and control signals, issuance of massages to the memory (RAM), transfer ofmessages through the output interface to the demodulator circuit and control of its behaviour.

If messages contained on an input control module message file are to be transferred, the input control module begins to, at prederterminated times, transfer the messages to the memory (RAM) through the output control module. When the three units of the memory have been loaded, the data transfer is interrupted between the modules until new commands are received. The output control module then starts the transfer of the information contained in the first unit of the mamory (RAM) to the demodulation circuit via the output interface. When all information from the first memory (RAM) has been transferred, the same process starts again for the information in the second and third memory units (RAM). However, when one of the memory units is unloaded, the output control module request the input control module to effect another data transfer. The reconstitution of the pulse train (binary signals) , that is transformation into analogic signals, is effected by the demodulator cicuit. Thus, all telephone line continous and uninterrupted playback/transmission of spoken messages is effected through the storage and reconstitution of the messages it the mass memory.

The equipment also comprises an additional unit called backgroud music circuit, which overlaps a continous musical tone and the spoken message being transmitted. This circuit ensures that, in the event of any input or output control module failure,a musical tone message remains on the line. The process of codification and decodification of analogic and digital-analogic signals is the same as previously described for spoken messages. However, the musical tone frequency sampling may be equal, higher or lower than used for spoken messages, wich ensures a better quality transmission. Moreover, the background music circuit has a microprocessor that controls all circuit inherent function independently from the input and output control modules. Thus, so that a telephone user may hear both the spoken and musical signal messages, a circuit is required for mixing the two types of messages and the exchange supervisory tone (N) ,while ensuring the user does not lose control over the call should any equipment failure occur.

Figure 2 shows the eletric diagram of the modulator circuit comprised of a low-pass filter IC7, Rl, R2, Cl and C2 and a modulator IC8. The analogic signal fed into input B is filtered by IC7, where the cutoff frequency is 3.4 KHz. IC8 modulates the signal at a frequency sampling rate of 32 KHz. Therefore, the modulator circuit transforms the analogic signal fed to input B into a serial digital signal at output B, wherefrom the signal is sent to the input interface. Frequency adjustments may be made as required due to distances and quality levels. Figure 3 shows the electric diagram of the demodulator circuit. The digital signal fed to input C is demodulated by IC9. The decodified signal (analogical signal) is delivered to IC10 by pin G (IC9) . IC10 comprises a low- pass filter with a cutoff frequency of 3.4 KHz and gain determinated by R18, R19. At output C, therefore, the analogical signal is a reconstituded message on its way to the mixing circuit.

Figure 4 shows the electric diagram of the interface comprised of a series parallel converter and a latch cicuit. The signal at input A is the voice digital signal coming from the delta modulator. IC1A and IC1B convert the input serial signal into an eight-bit parallel signal. The reading cycle by the parallel signal input control module is therefore 32/8 KHz, that is, 4 KHz. Thus, at each 250 Λ» sec, the control signal 1 is sent to the input control module, so that the signal reading of the converted signal be made through the issuance of signals SAZ and RD to allow IC2 to function through IC3A. Thereafter, the data is available in the data bus and can be read by the input control module.

Figure 5 shows the electric diagram of the output interface comprised of a parallel series converter and a latch circuit. Upon beginning the transmission of a message, the control signal 2 is sent to the monitoring unit of the output control module at each 250A*-sec. The search monitor unit then searches the memory (RAM) for a message sample, which, once found, remains available in the data bus. The SAl and WR signals cause the message samples to be sent, through CI5, from the data bus to the parallel series converter. CI 6 converts the eight-bit parallel signal into a serial signal at a frequency of 32 KHz. This analogic signal to output A, which will be reconstituted into an analogical signal by the demodulating circuit. Figure 6 shows the electric diagram of the monitoring unit of the output control module. The control unit is based on a micro-processor ( IC1 ) which operates at a 10 MHz crystal frequency, that is, a 5 MHz clock. Therefore, IC1 follows the program instructions and monitors the system, through the generation of all the command and monitoring signals required for the system to function perfectly. IC2 and IC3 have the function of a latch/buffer of the address bus. IC4 acts as the bi- directional buffer of the data bus. IC5, based on the

RD, WR and I/OM signals furnished by IC1, generates MR,

MW, I/OW and I/OR, respectively, for the memory devices and I/O installed in the system. These signals integrate the control bus. IC7, IC8, IC9 and IC10 are decodifying circuits wich, through the decodification of address lines in the address bus, generate the chip select signals for the rest of the system.

Figure 7 shows the electric diagram of the memory (RAM) . Each IC of the bank has an 8K x 8 bits organization. Thus, the total capacity of the bank is 258 Kbytes.

Access to the memories is possible through the monitor bus where the chip select (CS) signals are generated in the monitoring unit of the output control module. Figure 8 shows the electric diagram of the mixing circuit, which is constituted by three input amplifiers and a mixer. IC1, IC2, IC3 and associated components form the amplifier stage, the inputs being: channel 1, connected to the output of the demodulating circuit; channel 2, to the output of the backgroud music circuit; and channel 3, to the circuit of the tone generating machine of the exchange. The amplifiers serve the prupose of balancing the levels of the signals forming the signal sent to the telephone line. IC3 has the function of mixing the signals coming from the output of each amplifying circuit. Therefore, at output 5, the voice, background music and call control tone messages are mixed and the resulting signal sent to telephone address through exchanges.

Figure 9 shows the alternative block diagram of the equipment, which allows its part instalation in a telephone exhange, and other in any convenient place. Thus, the equipment part, up to the modulator circuit, may, if necessary, be instaled in other place than not a telephone exchange. The signals transmission for the second part of the equipment can be done by many means, like: FM Sub-channels (47, 64 or 92 KHz), UHF/VHF signals, phone lines, etc.

Claims

CLAIMS - "AUTOMATIC SYSTEM FOR THE PLAYBACK OF RECORDED SPOKEN

MESSAGES THROUGH A TELEPHONE LINE" - This system can either replace or be coupled to the tone and ring generating equipment currently in use at telephone exchanges is comprised of the following: input generating module, output generating module,modulating circuit, demodulating circuit, input interface, output interface, control unit, RAM memory, mixing circuit and background music circuit. It uses the time period normally assigned to the transmission of supervisory tones as well as other time periods during a telephone call also for transmitting spoken messages and background music concomitantly or not with the supervisory tones. - "AUTOMATIC SYSTEM FOR THE PALYBACK OF RECORDED SPOKEN

MESSAGES THROUGH A TELEPHONE LINE", as per Claim 1 above, is characterized by the fact that the modulating circuit is comprised of a low-pass filter, constituted by IC7, Rl, R2, Cl, C2 and IC8 modulator, which transforms analogic signals fed to input B into serial digital signals at output B and then to the input interface. - "AUTOMATIC SYSTEM FOR THE PLAYBACK OF RECORDED SPOKEN MESSAGES THROUGH A TELEPHONE LINE", as per Claim 1 above, is characterized by the fact that the demodulating circuit receives the digital signal fed to input C, where the signals are demodulated by IC9 and pass on to IC10 through Pin 6 of the said IC9_* - "AUTOMATIC SYSTEM FOR THE PLAYBACK OF RECORDED SPOKEN

MESSAGES THROUGH A TELEPHONE LINE", as per Claims 1 and 3 above, is characterized by the fact that IC10 is a low-pass filter with a cutoff frequency of 3.4 KHz and gain determinated by R18 and R19, the analogical signal being sent to the mixing circuit via output C. - "AUTOMATIC SYSTEM FOR THE PLAYBACK OF RECORDED SPOKEN

MESSAGES THROUGH A TELEPHONE LINE", as per Claim 1 above, is characterized by the fact that the input interface is comprised of a series parallel converter and a latch circuit, where digital voice signals coming from the Delta modulatr pass from input A to IC1A and IC1B, which convert them into eight-bit parallel signals which are then sent to the input monitoring module. - "AUTOMATIC SYSTEM FOR THE PLAYBACK OF RECORDED SPOKEN

MESSAGES THROUGH A TELEPHONE LINE", as per Claims 1 and 5, is characterized by the fact that the input monitor reads the converted signal through the issuance of signals SAZ and RD that allow IC2 to function through IC3A. - "AUTOMATIC SYSTEM FOR THE PLAYBACK OF RECORDED SPOKEN MESSAGES THROUGH A TELEPHONE LINE", as per Claim 1 is characterized by the fact that the output interface is comprised of a parallel series converter and a latch circuit that sends the messages to the control unit of the output monitoring module via Control 2 at each 250 M-sec. - "AUTOMATIC SYSTEM FOR THE PLAYBACK OF RECORDED SPOKEN

MESSAGES THROUGH A TELEPHONE LINE", as per Claims 1 and 7, is characterized by the fact that the control unit of the output monitoring module comprises an ICl microprocessor that, operating at a 10 MHz crystal frequency, follows program instructions and monitors the system. - "AUTOMATIC SYSTEM FOR THE PLAYBACK OF RECORDED SPOKEN

MESSAGES THROUGH A TELEPHONE LINE", as per Claim 1, is characterized by the fact that the mixing circuit is comprised of three input amplifiers and a mixer, where ICl, IC2 and IC 3, as well as all related components, form the amplifying stage. - "AUTOMATIC SYSTEM FORTHE PLAYBACK OF RECORDED SPOKEN MESSAGES THROUGH A TELEPHONE LINE", as per Claims 1 and 9 , is characterized by the fact that the amplifying stage inputs have channel 1 connected to the demodulating circut output , channel 2 to the background music circuit output and channel 3 to the tone generating machine of the telephone exchange.