Abstract: Systems and methods for self-administering a sound test to verify operation of a speaker and/or alarm within a hazard detection system are described herein. The sound test can verify that the audible sources such as the alarm and speaker operate at the requisite loudness and frequencies. In addition, the sound test can be self-administered in that it does not require the presence of a person to initiate or verify that the audible sources are functioning properly.

Abstract: A system and method for processing a plurality of channels, for example audio channels, in parallel is provided. For example, a plurality of telephony channels are processed in order to detect and respond to call progress tones. The channels may be processed according to a common transform algorithm. Advantageously, a massively parallel architecture is employed, in which operations on many channels are synchronized, to achieve a high efficiency parallel processing environment. The parallel processor may be situated on a data bus, separate from a main general purpose processor, or integrated with the processor in a common board or integrated device. All, or a portion of a speech processing algorithm may also be performed in a massively parallel manner.

Abstract: In one aspect thereof the exemplary embodiments of this invention provide a method that includes, prior to confirmation that a network access node has correctly acquired capabilities of a user equipment, operating a user equipment with the network access node in accordance with a first rate matching mode and, only after confirmation to the user equipment that the network access node has correctly acquired capabilities of the user equipment, changing the rate matching mode to a second rate matching mode. In an embodiment the first rate matching mode comprises puncturing a downlink shared channel transmission with a set of resource elements, which may be at least one of reference symbols and muted resource elements, and the second rate matching mode comprises rate matching the downlink shared channel around those resource elements that are members of the set of resource elements. Also described are apparatus and computer readable storage medium storing program code that operate in accordance with the method.

Abstract: A system and method for processing a plurality of channels, for example audio channels, in parallel is provided. For example, a plurality of telephony channels are processed in order to detect and respond to call progress tones. The channels may be processed according to a common transform algorithm. Advantageously, a massively parallel architecture is employed, in which operations on many channels are synchronized, to achieve a high efficiency parallel processing environment. The parallel processor may be situated on a data bus, separate from a main general purpose processor, or integrated with the processor in a common board or integrated device. All, or a portion of a speech processing algorithm may also be performed in a massively parallel manner.

Abstract: Methods, systems and apparatuses can be used to detect and prevent invalid or unwanted pulses. In various implementations, pulse information received from a telephony device can be used to identify a duration associated with the pulse. Based upon the duration, a determination can be made whether the pulse is invalid.

Abstract: Methods, systems and apparatuses can be used to detect and prevent invalid or unwanted pulses. In various implementations, a series of dial pulses received from a telephony device can be counted to determine whether the series is valid. Based upon the count, a determination can be made whether the series of dial pulses is invalid.

Abstract: Methods, systems and apparatuses can be used to detect and prevent invalid or unwanted pulses. In various implementations, pulse information received from a telephony device can be used to identify a duration associated with the pulse. Based upon the duration, a determination can be made whether the pulse is invalid.

Abstract: A method and apparatus for switching active calls from one entity to another on a network device. The method includes the steps of collecting the call information, designating a second, or target, entity to receive the call, and switching the call. If compression is being used in the call, the method will include the steps of copying the compression dictionary tables and loading the tables in the second entity. The apparatus includes a controller and at least two entities. A connector connects the entity with incoming or outgoing phone calls. The controller switches the active call from a first entity to a second entity.

Abstract: Stereo sound signals are reproduced directly from loudspeakers (SP(L), SP(R)). By using a sum signal and a difference signal of the stereo sound signals as a reference signal, and according to a cross spectrum calculation of the reference signal with a microphone-collected sound signal, calculation is performed to obtain transfer functions of four sound transfer systems between the loudspeakers (SP(L), SP(R)) and microphones (MC(L), MC(R)). The transfer functions obtained are subjected to inverse Fourier transform to obtain impulse responses, which are set in filter means (40-1 to 40-4) to create echo cancel signals and perform echo canceling. This solves the problem of an indefinite coefficient in the echo cancel technique of a multi-channel sound signal.

Abstract: A method is set forth for reducing the total acoustic echo cancellation convergence time for all look directions in a microphone array based full-duplex system. The method is based on capturing the loudspeaker signal due to the first far-end speech bursts when the conferencing system is first used, as well as the corresponding loudspeaker feedback signals in the individual microphones. The captured signals are then used for consecutive adaptation of the acoustic echo canceller on all echo paths corresponding to all look directions of the beamformer, thereby training the AEC. This training process can be executed concurrently with normal phone operation, for example, as a background process that utilizes available processing cycles.

Abstract: A multimode optical fibre communications system, and in particular to a system in which non-linearities in the propagation of the signal through a multimode optical communications channel degrade the signal presented to the receiver. The system includes an optical transmitter unit for connection to a multimode optical fibre transmission link. The transmitter unit has a data input for receiving an input data signal, a data signal processing circuit and a source of optical radiation. The data signal processing circuit is arranged to receive the input data signal from the data input and to provide a processed data signal to the source of optical radiation and the source of optical radiation is arranged to generate from this an optical signal for transmission by a multimode optical fibre.

Abstract: A method is provided for supporting an enhanced dialing service in telecommunications equipment. The method includes: receiving a series of dialed digits from customer premises equipment (CPE) (10); determining if the dialed digits correspond to a designated identity; as each dialed digit in the series is received, advancing a call processing state of the telecommunications equipment in accordance with the dialed digit when the dialed digit does not correspond to the designated identity; and, returning the call processing state of the telecommunications equipment to a prior call processing state when the dialed digit does correspond to the designated identity.

Abstract: A tone transmission part transmits a tone signal. A tone signal detection part detects data representing an echo signal of the tone signal transmitted by the tone transmission part, and detects data representing a predetermined tone signal targeted to be detected. The tone signal detection part also detects data representing tone signals of all of input signals. A tone judgment part excludes the data representing the echo signal from the data representing the tone signals of all of the input signals so as to judge a detection of the predetermined tone signal.

Abstract: A push-button signal receiving circuit and a push-button signal detection method are provided which can enhance frequency detection accuracy and yet can reduce erroneous determination as to the valid length of PB signal and thus erroneous operation. Valid signal determining portion determines the validity of a received PB signal in accordance with the results of determination as to the coincidence of continuance times of identification signals extracted by first and second frequency detecting portion for performing frequency detection over a small number of periods, and the coincidence of the frequencies detected by the first and second frequency detecting portion with those detected by third and fourth frequency detecting portion over a large number of periods.

Abstract: The present invention provides a low power method and apparatus for detecting a signaling tone. The apparatus comprises a resonator which can be configured to resonate at a specified frequency, a controller for configuring the resonator, and a comparator for comparing the output of the resonator as configured by the controller. The method comprises configuring the resonator to resonate at a first frequency to generate an output, configuring the resonator to resonate at a second frequency to generate an output, comparing the outputs, and indicating the presence of a signal if the output of the resonator configured to resonate at a first frequency exceeds the output of the resonator configured to resonate at a second frequency by a predefined amount.

Abstract: An off-hook management telecommunication system comprising a DTMF detector, an encoder and a microprocessor located within a channel unit. The system allows the user to depress a selected sequence of keys during a line connection which indicate to the microprocessor to hold the line off-hook, via signaling to a central office, despite the fact that the telephone is being returned to an on-hook position. The line is held off-hook for a predetermined amount of time to allow the user to switch telephones.

Abstract: System and methods for determining an optimized transmit spectra (spectral distributions of transmission power) for a set of communications channels that experience cross-talk among themselves and for transmitting data on the channels. The transmit spectra are preferably constructed so that largely contiguous frequency bands are allocated to each signaling direction (upstream/downstream) on each communications channel and/or to each channel in the set of channels. In one embodiment, each communications channel is restricted to a maximum time-averaged power. The method preferably includes steps of determining the channel transfer functions of the communications channel, determining interference characteristics of the channels, calculating substantially optimal transmit spectra for the communications channels, and redistributing the frequency bins so that they are contiguously grouped in each transmit spectra. The contiguous groupings allow wider frequency bands for signaling in the channel.

Abstract: An information processing apparatus has an SDL execution unit described in a specification and description language (SDL), an external environment description unit described in a programming language other than the SDL, and an adapter having a pseudo internal signal generator and a queue manager. A task of an operation system is assigned to the external environment description unit, which is combined with the SDL execution unit through the adapter. The pseudo internal signal generator converts signals including messages, events, and parameters to be transferred between the SDL execution unit and the external environment description unit. The converted signals are transferred between the SDL execution unit and the external environment description unit through the queue manager. Also provided is an information processing method to materialize the information processing apparatus. These method and apparatus reduce the overhead of the operation system.

Abstract: A system and method for DTMF tone detection receives an input signal. The input signal includes a DTMF tone. The DTMF tone has a first frequency and a second frequency. An initial first frequency and an initial second frequency of said DTMF tone are selected by calculating a plurality of cost functions. The initial first frequency is confirmed to be the first frequency and the initial second frequency is confirmed to be the second frequency using re-computed values of the plurality of cost functions.

Abstract: A system and method for allowing callers using DP telephones to access IVR systems, including monitoring DP signals until a dialed digit is recognized, where the DP signals are generated by a transmission element and received by a reception element within a telephone network, and where the monitoring does not interfere with the DP signals, sending the dialed digit to a DTMF transmitter, and outputting a DTMF signal following the DP signals on a speech path, wherein the DTMF signal corresponds to the dialed digit.

Abstract: A line card integrates subscriber line interface circuitry, A/D and D/A converters, and digital signal processing technology. The digital signal processing technology performs many line card tasks such as switch hook detection, ground key detection, DC feed control, polarity reversal, ringing tests, fault detection, power cross detection, and ring trip detection. The digitally processed DC feed control ensures that digitally filtered A (Tip) and B (Ring) conductor voltages closely follow actual subscriber loop Vab voltages to avoid circuitry saturation and to validate power feed operating point calculations relative to actual subscriber loop conditions. Furthermore, the DC feed control opens a Vab fluctuation window in high current gain areas of subscriber loop power feed to reduce noise.

Abstract: An input digital sample containing voice and DTMF signal is used as an input to an adaptive filter. The output signal of the adaptive filter is used to generate a residue from the difference between the input sample and the output signal. The residue signal is used to update the coefficients of the adaptive filter. Upon the convergence of the adaptive filter, the residue signal comprises voice reduced DTMF signal while the adaptive filter's output signal contains the detected DTMF signal.

Abstract: A technique for minimizing the number of users which must dial training digits in a voice processing system. The system uses the first digit normally dialed as part of the user responses to the voice processing application, and measures such digit to determine whether it and future digits can be detected accurately without the use of a training digit. If they can, no training digit is required. If they can't, then a training digit is required, and the user is prompted for the same.

Abstract: A dial pulse detector in a paging system for rendering a paging service for a subscriber who uses a mechanical telephone includes a delay correlativity calculator for receiving dial pulses generated by predetermined pre-test dialing digits dialed by the subscriber, calculating the correlativity between the delayed dial pulse and undelayed dial pulse to sample a reference signal of the dial pulse. An energy correlativity calculator calculates the energy correlativity between the sampled reference signal and the successively received dial pulse to examine the validity of the reference signal. A dial pulse verifier operates according to recognition of the validity of the reference signal by the energy correlativity calculator and checks input number of times and input time of the dial pulse so as to verify whether the dial pulse corresponds to the pre-test dialing digits.

Abstract: The present invention is related to a telecommunication system and method enabling a user to get access to any of automated call processing systems. The system comprises in combination a telephone line interface means adapted to provide function of a telephone set, a control means connected to the telephone line interface means, and adapted to constantly monitor the telephone line, analyze a call progress and a ring back tone signals and to switch automatically without human intervention from a pulse dialling mode to a DTMF dialling mode upon detection of a ring back signal validated by a central office. This system also includes a handset and keypad means connected to a telephone interface means and the control means. The central office processes calls in a pause dialling mode. The system enables the user to get access to automated call processing systems, the access to which is limited only to callers equipped with a DTMF telephones and central office base DTMF services.

Abstract: An impulse convertor for converting a received telephone impulse dialing signal into a restored dialing signal has a limiter-amplifier, a signal detector, an output formatter, and a controller. The limiter-amplifier receives the impulse dialing signal from an interface b to a communication system such as a public switched telephone network or a cellular network, limits the amplitude range of the received impulse dialing signal, and compensates for attenuation of the received impulse dialing signal. The signal detector analyzes successive portions of an output signal generated by the limiter-amplifier and determines dialed digits represented by the received impulse dialing signal. The output formatter converts the dialed digits determined by the signal detector into the restored dialing signal that is provided to a device C such as a PABX. The controller coordinates operation of the limiter-amplifier, signal detector, and output formatter.

Abstract: A method and system for efficiently recognizing the last digit of a computer-dialed DTMF telephone number. The actual gaps between a number of first-dialed successive digits are compared with a standard gap exhibited during computer dialing to determine whether the dialing in progress is manual or automatic. As dialing proceeds, the absence of a digit for a preset time indicates that dialing has been completed. The preset time is set to a small value if it is determined that computer dialing is in progress.

Abstract: A telephone branch line transmission circuit includes a control device which receives dial pulses and in response thereto controls at least two switching devices arranged between two branch lines to a telephone exchange. A blocking capacitor is coupled to the first switching device. In order to prevent charging of such capacitor from distorting the current supplied from the exchange at the end of a dial pulse, the first switching device is included between the blocking capacitor and one of the branch lines. The connection node thereof with the blocking capacitor is connected to a terminal of a second switching device, another terminal of which is coupled to the second branch line. The control terminal of the second switching device is connected to a storage capacitor. In the off-state of the switching devices no residual direct current can flow from the exchange to charge the blocking capacitor.

Abstract: Method and apparatus for detecting and identifying dial pulse digits in a telephone signal. An embodiment of the inventive method includes the steps of: detecting a dial pulse "0" digit; analyzing the dial pulse "0" digit to determine a predetermined set of data to provide a template which is characteristic of the dial pulse "0" digit; detecting a further dial pulse digit; analyzing the further dial pulse digit to determine a predetermined set of data which is characteristic of the further dial pulse digit; and comparing the predetermined set of data of the further dial pulse digit with the template to identify the further dial pulse digit.

Abstract: Device for connecting one out of a plurality of internal lines (34,36) to an information exchange subscriber line, such as a telephone, facsimile or telex line, provided with decoding means (28,1-16) to detect dialling disc pulses (a), preferably on base of the time period between edges (b',b") of these pulses, timing means (9,10,12) to determine a time interval after an edge in which the next edge should occur and a test means (13) to ascertain whether said next edge occurred indeed in said interval.

Abstract: Apparatus and method for receiving and decoding the bursts of line current activity produced by rotary dial action in dialing a sequence of digits. In accordance with the apparatus of the present invention, a rotary dial pulse receiver counts the number of bursts of line current activity produced by a given telephone handset in dialing a standard test digit. A conversion means selects a conversion algorithm which is effective to express the relationship between the number of bursts counted and the value of the digit dialed and applies the selected algorithm to decode the value of subsequent non-standard digits dialed. In accordance with the method of the present invention, a method of converting the number of bursts in line current activity produced through rotary dial action in dialing a sequence of digits by selecting and applying a conversion algorithm is described.

Abstract: A dial pulse detection system 20 has an analogue input buffer 22, an analogue to digital converter (ADC) 24, a processor 25, and an output buffer 26. Processor 25 operates on the digitally encoded samples produced by ADC 24 from the received signal to achieve detection of the dial pulse signalling data in the received signal. This detection process involves a training process and a recognition process. The training process extracts amplitude and timing characteristics of a first part of the received signal from a dialling instrument. These characteristics are then used to form a template for recognition of subsequent dialled digits. Detected digits are then made available at output buffer 26. Processor 25 includes a facility for sending an error signal to the user to indicate that dialled digits have not been correctly received.

Abstract: A signal persistence time interval recognition system reproduces a condition of an input signal as a condition of an output signal only when the input signal condition persists for at least a predetermined time interval. The system periodically scans the input signal condition and has a first memory (ROM) for storing a start value indicative of the time interval. A second memory (RAM) stores a value indicative of the time counted since the detection of a difference between the input and output signal conditions. A third memory stores the output signal condition and a processor brings the start value from the first memory into the second memory when no difference is detected. The processor modifies the value in the second memory each time such a difference is detected and until a value is reached indicating that the time interval has been counted, the output signal condition in the third memory being then changed.

Abstract: A multiprocessor system includes a plurality of signal processors and a common unit processor. Each of the signal processors is connected to a different source of signals such as voice signals and performs one or more signal processing functions relative to the connected source. The common unit processor performs one or more functions for the signal processors on a shared synchronized basis. A signal processor adapter responsive to a source of clock pulses generates synchronous interrupts applied to the common unit processor and enabling signals in sequence to connect the signal processor in sequence to the common unit processor in synchronization with the interrupts. In addition, the signal processors are provided with ping-pong buffers at their inputs and outputs to enhance throughput.

Abstract: A method is disclosed for measuring telephone dial pulse parameters independent of telephone subscriber line length. To avoid distortions resulting from varying subscriber line lengths, there is performed during the first pulse interval of a dial pulse series to be measured, a voltage measurement with results in a value that depends on the line length, and, in accordance with this value, the times measured for pulse interval and pulse duration of the dial pulse series are corrected.

Abstract: A direct inward dial integration device is connected to the trunk side of a PBX in a business telephone system. The present invention permits a voice message system (VMS) to answer forwarded calls with a personal greeting of a called party. When a direct inward dial call is made, the identity of the extension (i.e., the extension number) being called is transmitted to the PBX by the telephone system. The integration device detects and stores the extension identity. If an incoming call is forwarded to the VMS, the VMS generates a query sigal, which causes the integration device to automatically transmit the stored extension number to the voice message system. Thus the voice message system can identify the called party. The VMS responds by answering the call with a personal greeting of the called party and inviting the outside caller to leave a recorded message.

Abstract: A pulse receiver for connection in a telephone circuit including two coils connected between a voltage potential and ground further including an external impedance that is alternately connected between the coils. The pulse receiver includes a first impedance connected in series with the first coil and a second impedance equivalent to the first impedance connected in series with the second coil and an input circuit connected across the first impedance for detecting the connection of the external impedance of the telephone circuit. Switching circuitry is provided that is connected to the input circuitry and alters a first output node pair from a normally closed state to an open state and alters a second output pair from a normally open state to a closed state in response to a signal received from the input circuitry indicative of the detection of the external impedance connection.

Abstract: The present invention is an improved subscriber line interface circuit which allows fast detection of an off-hook signal in the presence of a ringing signal during an answer mode while also permitting fast detection of dialing pulses during a calling mode. A programmable filter is used in the supervision circuit of the SLIC to allow the cutoff frequency of the filter to be varied so that the 20 Hz ringing signal will be attenuated during a ringing sequence and dialing pulse rates up to 20 Hz will be passed by the filter during the calling mode. A clamping amplifier is used to clamp the received signal to a maximum of 1.5 times the loop threshold current. This eliminates the large variations in the rise and fall times of the pulse dialing signal due to variations in the loop current caused by varying impedances of the telephone line. The filter is programmed by using an analog switch to bypass certain filter elements.