Abstract: A technique for eliminating tone signal noise generated using a speakerphone, in which the communication path of a microphone is interrupted when a key is pressed, and then the communication path of the microphone is linked only after a constant time period after the key has been released, thereby eliminating the noise input through the microphone during usage of the speakerphone.

Abstract: Type 2 caller ID/call waiting (CIDCW) customer premises equipment (CPE) is provided which has a greatly improved method of detecting tones indicating the availability of call information regarding an incoming call while the customer premises equipment is in an off-hook condition. A hybrid echo canceler (HEC) algorithm is added to the customer premises equipment (CPE) to suppress the signal from the microphone of the customer premises equipment from being included in the signal analyzed for the presence of tones. Thus, the microphone of the customer premises equipment need not be muted upon detection of the alerting CAS tone sequence in a Caller ID service. The HEC algorithm runs substantially continuously in a preferred embodiment, and may be combined with a conventionally operating HEC which cancels reflections due to the hybrid or telephone line interface.

Abstract: A voice switch in a hands-free communications system performs selective attenuation with respect to respective voice signals being transmitted and received in respective transmitting and receiving paths, the transmitted voice signal having been converted from an audible voice by a microphone connected to the transmitting path and the received voice signal having been converted by a loudspeaker to an audible voice output. A detector selectively detects a currently transmitted voice signal at a normal level, a currently received voice signal at a normal level, and drops in the respective levels thereof to nil levels and provides corresponding outputs to a controller. When one of the transmitted and received voice signals is of a normal level and the detector newly detects the other thereof at a normal level, the controller selectively attenuates that other, newly-detected normal level voice signal.

Abstract: A telephone circuit is composed of a current circuit, an hybrid circuit, a noise inhibiting unit, and a CAS detector circuit. The current circuit has two input ends connected with tip and ring of the telephone line, and two output ends connected with two input ends of the hybrid circuit. The signal received by two input ends of the hybrid circuit is analyzed. The noise inhibiting unit is connected in series between the hybrid circuit and the CAS detector circuit and is provided with at least one inhibiting circuit which has two input ends connected with the first signal line and the second signal line of the hybrid circuit. The output ends of the inhibiting circuit are connected with two input ends of the CAS detector circuit for minimizing the error of identifying the CAS signal by the CAS detector circuit.

Abstract: A wireless terminal is disclosed comprising a first electro-acoustic transducer for converting a first acoustic signal to an uplink signal; a wireless transmitter capable of transmitting the uplink signal to a base station; a wireless receiver capable of receiving a downlink signal from the base station; a terminal processor for conditionally adding a portion of the uplink signal to the downlink signal while a first parameter is set and not adding the portion of the uplink signal to the downlink signal while the first parameter is reset; and a second electro-acoustic transducer for converting the downlink signal to a second acoustic signal. Another embodiment is disclosed that reduces the sidetone level of the terminal based on the power level of the envelope of the acoustic signal input at the microphone.

Abstract: A telephone apparatus is connected to a plurality of telephone lines and has a transfer function for transferring a call among the plurality of telephone lines. The telephone apparatus includes circuit interfaces each of which has a 2-4 line substitution function and conducts an interface function between each of the plurality of telephone lines and a telephone apparatus internal circuit. A first amplifier amplifies a signal from a first telephone line and a second amplifier amplifies a signal from a second telephone line. A control unit compares the levels of the signals from the first and second telephone lines to control the amplification factors of the first and second amplifiers on the basis of a result of the comparison.

Abstract: In an apparatus of adjusting a sound level of a reception voice in a telephone, an audio signal including a transmission voice signal and a peripheral noise is inputted through a microphone. A gain control signal corresponding to a level of the peripheral noise is generated while the transmission voice signal is not present and the gain control signal immediately before input of the transmission voice signal is started is held and outputted while the transmission voice signal is present. A variable gain amplifier amplifies a reception voice signal in accordance with the gain control signal such that a voice corresponding to the reception voice signal is outputted from a speaker with a sound level determined in accordance with the gain control signal.

Abstract: The present invention overcomes interface problems between proprietary handset ports on telephone base units and voice/data accessory products by allowing a user to automatically calibrate the telephone accessory product for an optimal interface match with the intended telephone base unit. This is accomplished through the use of a "Smart Interface Technology" (SIT) integrated chip set consisting of a full custom analog and semi-custom digital integrated circuit. The SIT incorporates three different methods for "learning " the characteristics of 4-wire port modular interfaces found in all telephone station sets. Basically, these methods determine the appropriate 4-wire terminal configurations, the transmit and receive channels of the intended telephone base unit, and adjust the channel sensitivities until an optimal and clear signal is provided for the user.

Abstract: A telephone handset interface reduces or eliminates feedback induced singing in a telephone handset/hybrid loop. The interface provides for user selectable volume control settings, each setting representing an amount of gain or loss to be applied to the transmit and receive signals within the telephone set. The gain settings are selected to limit overall loop gain so as to reduce or eliminate any tendency of the handset to sing at high receive gain levels. Loss inserted into the transmit circuit is removed when user speech is detected in the transmit circuit.

Abstract: An acoustic feedback suppression system, particularly for auxiliary ringers in socket telephone systems, comprising a first terminal and a second terminal of a telephone line, an internal ringer, and an auxiliary ringer that is connected between the first and second telephone lines. The system comprises an activation circuit adapted to activate first and second switching circuits to reduce the voltage at the terminals of the auxiliary ringer, the activation circuit being controlled by an activation signal generated by a control and signaling circuit when a user dials a telephone number.

Abstract: The echo canceler includes an N-tap FIR filter and a coefficient controller. Previous filter coefficients which were generated before an estimation error occurrence when the transmitting output signal becomes greater in power than the transmitting input signal are obtained by using only a number S of coefficient correction values and another number (N-1+S) of sequentially received signals. Since the latest (N-1) received signals are stored in (N-1) delay circuits of an N-tap FIR filter, respectively, a relatively small capacity memory may be provided to store the earlier S received signals. The number S is determined so that a time interval determined by the number S of sampling instants is longer than a response time of a double talk detector.

Abstract: A balanced network in a keyphone system has a filtering unit which is composed of: first and second filtering circuits, formed on the transmission path and having an inverted-phase feature, for adjusting the level of the transmission signal and for removing spurious waves thereof; an inverted-phase signal branch point disposed between the first and second filtering circuits, for branching out an inverted-phase transmission signal; an operational amplifier disposed on the reception path, for adjusting a level of the reception signal; a signal superimposing node disposed ahead of an input of the operational amplifier; and a feedback signal level adjusting circuit disposed between the inverted-phase signal branch point and the signal superimposing node, for adjusting the level of the inverted-phase transmission signal which has been branched out from the inverted-phase signal branch point so as to reduce a feedback signal, transmitted from the transmission path to the reception path through the common node, to a

Abstract: The invention provides an apparatus and method and a device for echo attenuation in a telephone. According to the invention, the echo attenuation in the telephone is adapted to the background noise such that the ambient background noise in telephone (10) is measured with a noise detector (1) and the noise level is determined in a background noise measurement unit (2). The background noise that has been determined is used to set the attenuation parameters in parameter determining unit (3). The signal level received from the other subscriber's telephone (11) is measured in input signal level measurement unit (4). The feed signal of the telephone is attenuated in attenuation Control unit (6) on the basis of the attenuation values calculated, which have been determined from the attenuation parameters and the signal level received by the other subscriber in output signal attenuation determining unit (5).

Abstract: The speech circuit matches the impedance of the telephone line by synthesizing a complex impedance using a positive feedback loop which has a single resistor (11), and cancels out the side tone using a subtractor (20') which extracts from the signal (Va) coming from the line a signal (Vb) correlated to the signal to be transmitted. In order to achieve cancellation of the side tone unaffected by the noise produced in the impedance synthesizing circuits, the signal (Vb) is derived by processing the signal present in the resistor (11) at the output of the feedback loop.

Abstract: A cordless telephone has receive (12) and transmit (14) signal paths for passing voice signals. Sidetones normally appear in the receive signal path from the near party's voice. A signal strength comparator (34) monitors the transmit signal path and the receive signal path and asserts a gain control signal when the transmit path signal strength exceeds a threshold set to a predetermined value below the receive path signal strength. The gain control signal decreases the gain (42) in the receive signal path to reduce undesirably loud sidetones in the speaker earpiece. When the transmit path signal strength is less than the predetermined threshold, the gain control signal is not asserted allowing maximum amplification in the receive signal path as the sidetone is sufficiently small as to not interfere with the main received voice signal, or otherwise become noticeably loud in the speaker earpiece.

Abstract: A mobile radio terminal comprises a speech processor for processing a first and at least a further speech signal formed by noise and speech signal components and available as sample values. The sampled further speech signal is delayed by an adjustable delay value. Control means are provided which are used for forming gradient estimates. The control means are additionally used for recursively determining delay estimates from the gradient estimates. By rounding the delay estimates, the delay values are formed. Furthermore, their mutually time-shifted speech signals are added together by means of an adder device.

Abstract: A speech circuit is disclosed which solves the serious problem of the degradation of the articulation of received speech voice in conventional circuits and permits pleasant communications at places where the background noise level is high. The circuit has a construction in which an input signal from a microphone is attenuated in correspondence to the background noise level to form a sidetone signal and a received speech signal from a speech channel is amplified in correspondence to the background noise level to form a new received speech signal.

Abstract: A telephone set comprises a noise detector, a noise level determining circuit, and a noise canceler. The noise detector detects a noise component from a transmitting acoustic signal by detecting signal recurrence of the transmitting acoustic signal. The noise level determining circuit determines a noise level by comparing the noise component with a first threshold. According to the noise level, the noise canoelet cancels the noise component from the transmitting acoustic signal when the noise level is not smaller than the first threshold. The telephone set further comprises a frequency characteristics controller for a received acoustic signal to obtain a desired speech articulation. The frequency characteristics controller, based on the noise level, determines a level change region, a level change value, and a level change direction of the spectrum of the received acoustic signal.

Abstract: A speaker amplification system incorporates an adaptable notch filter that can dynamically adapt to the frequency of feedback oscillations caused by acoustic feedback frequency in order to remove the feedback oscillation before they are amplified above an audible level. The adaptable notch filter is implemented as an adaptable finite impulse response filter by means of a digital signal processor. The finite impulse response filter also has an adaptable convergence factor which minimizes noise introduced by the filter. A special limiting routine is used to limit the values of the finite impulse response filter coefficients to prevent the filter from "blowing up" due to large input signal disturbances. The speaker amplification system also includes circuitry which provides multiple bands of equalization, with each band being capable of amplification.

Abstract: An audio caller identification unit is connected between a user's telephone and a telephone exchange providing calling party identification, such as calling number or name. The unit has user accessible audio inputs, either a microphone or an interface to a handset of the telephone. The unit also has audio message memory, a user programmable table of concordance for tacking the location of audio messages, a record/playback device, a speaker, and a controller. The unit may also have a visual display. Control keys may be connected to the controller, however their function may also be performed by using a keypad of the telephone. A speech generator may also be connected to the controller and to the speaker. In operation, the unit has programming, review, and monitoring modes.

Abstract: To improve the anti-sidetone of electro-acoustic installations, particularly to improve the intercom characteristics of a telecommunication terminal, with compander control by an echo evaluation circuit (52) assigned to the compander (4), the latter has an echo compensator (40) located upstream, and/or a voice-controlled balance (51) is located downstream of the compander (4), which are controlled by the echo evaluation circuit (52) of the compander (4). To control the echo compensator (40) and/or the voice-controlled balance (51), the usual control circuitry required for their operation can be omitted. The result is a particularly effective, economical method with very good intercom characteristics.

Abstract: Selectively compressing the signal supplied to a loudspeaker based upon the volume setting for the sound level to be produced by the loudspeaker improves the intelligibility of human speech produced by a loudspeaker at high volume levels. Compression of the signal to the loudspeaker prevents the loudspeaker from overloading. The level of compression may be adjusted based upon the volume setting. The amplitude of an audio signal sent to a loudspeaker may be increased if below a preselected pivot point and increased if above the pivot point. The amount of increase or decrease in the amplitude may be proportional to the difference between the amplitude of the audio signal and the pivot point. This difference may also be raised to the power of a shape factor to determine the amount of increase or decrease in the amplitude of the audio signal.

Abstract: An attended messaging system having an audio message recording and playback device connected in series between a telephone base unit and a telephone receiver handset or headset. The recording and playback device includes a solid-state audio memory chip for storing a message and pairs of operational amplifiers connecting the memory chip of the recoding and playback device with the telephone base unit and with the speaker and microphone of the telephone receiver handset or headset in a differential circuit of audio signal channels. The recording and playback device is manually-actuable by means of buttons or switches to operate in a recording mode, in which a message spoken into the handset or headset microphone is stored in the audio memory chip, or in a playback mode, in which the previously recorded message is communicated through the telephone base unit and telephone line to a designated remote telephone station with which audio communication has been established by the telephone base unit.

Abstract: Our audio processing system for use in teleconferencing systems includes a microphone, receiver, and substantially the same comb filter and frequency scaler at each station for reducing acoustic echo and increasing the margin of acoustic stability. The comb filters and frequency scalers cooperate with one another in that the frequency scalers each induces a shift in frequency which corresponds to the peak-to-trough spacing of the transfer function of the comb filters. Our audio processing system may also include auxiliary echo suppressors for reducing acoustic echo and increasing the margin of acoustic stability in systems having high and low transmission delays. Other devices may be employed for reducing noise and smoothing out the operation of individual components in the system.

Abstract: Speech signals appearing at the receiving and transmitting paths of a speakerphone which operates under voice switching, are received at a control circuit. The control circuit issues a control signal which assumes a first level in the event that a signal at the receiving path exceeds a signal at the transmitting path and which assumes a second level if the signal at the receiving path falls below the signal at the transmitting path. Communication loss is inserted in both the receiving and transmitting paths in response to the first level of the control signal, and is eliminated therefrom in response to the second level of the control signal.

Abstract: A received signal vector x(n), a coefficient error covariance matrix P(n) from a memory part 12 and a forgetting factor .nu. from a memory part 13 are provided to a gain calculating part 14 to obtain a gain vector k(n). The thus obtained gain vector k(n) and an error e(n) between an echo and an echo replica are multiplied in a multiplying part 16. The multiplied output and a filter coefficient h(n) from a memory part 18 are added together to update the latter. The thus updated filter coefficient is used as the filter coefficient of an estimated echo path (an FIR filter, for example). The coefficient error covariance matrix P(n), the gain vector k(n), the received signal vector x(n) and the forgetting factor .nu.

Abstract: A telephone handset outbound noise canceller uses an omnidirectional reference microphone placed near the transmitter grille. Two adaptive filters are employed, one driven by the handset transmitter to subtract speech from a reference value to produce an enhanced reference signal; and a second adaptive filter driven by the enhanced reference signal to subtract noise from the transmitter. The weights of the two adaptive filters are allowed to adapt or are frozen according to criteria set as a result of transmitter power measurements which determine whether speech is likely to be present or not.

Abstract: A telephone set includes an anti-sidetone circuit formed by a feedback loop having a filter in the feedback path. The feedback loop output is substantially equal to the signal produced at the set connection terminals by the microphone. Preferably the filter is composed of two sub-filters connected in cascade, formed of switched capacitors in an integrated circuit. An EEPROM control circuit can be included to permit easy programming during manufacture, to match a selected transmission line impedance.

Abstract: An echo canceller for use in send/receive apparatus includes a frequency-domain adaptive filter which provides filter parameters to a time-domain programmable filter. The programmable filter produces a signal y(k) which is an estimate of the echo signal produced by acoustic coupling between a loudspeaker in the received path and a microphone in the send path. By subtracting this estimate from the signal produced in the send path, an output signal r(k) is obtained from the send path which is essentially free of echoes. An accurate estimate of the echo signal cannot be obtained if a send signal is present at the input of the send path. Therefore, detecting means in the adaptive filter detects whether such a send signal is present and in that case blocks the transfer of filter parameters from the adaptive filter to the programmable filter. Erroneous modification of the estimated echo signal is thereby avoided.

Abstract: An audio processing system for use in teleconferencing systems having high and low transmission delays. The audio processing system utilizes complementary comb filters in combination with auxiliary echo suppressors, a frequency scaler, expander/gates and automatic level controlling devices to substantially improve acoustic stability margin and reduce far-end talker echo.

Abstract: An adaptive echo cancellation apparatus disclosed herein has three channels and a channel selection circuit. The first channel serves to block an outgoing digital signal from flowing to a transceiver. The second channel is used to directly couple the outgoing digital signal to the transceiver. The third channel serves to subtract a predicted echo signal, estimated by using the incoming digital signal, from the outgoing digital signal and to couple an error signal, representative of the difference therebetween, to the transceiver. The adaptive channel selection circuit is used to select one of the channels depending on the presence of the valid outgoing and incoming digital signals in order to effectively reduce the estimation operation. Further, the third channel is provided with a filter wherein the filter coefficients are adaptively updated by using the signs of a predetermined number of samples, thereby achieving a considerable reduction in the computational complexity.

Abstract: An audio processing system (100) for use in a teleconferencing system (10) utilizes complementary comb filters (42, 52, 220, 320) in combination with an echo suppressor (210, 310) and/or a frequency scaler (60, 120) to substantially improve acoustic stability margin and reduce far-end talker echo.

Abstract: The process of the present invention operates on the voice codewords transmitted in a digital cellular radiotelephone environment. The process first determines if a voice is active (201) on the communications path from the base station to the radiotelephone. If a voice is present, a threshold is calculated (202). Any signals on the reverse path from the radiotelephone to the base station that are below this threshold (203) are replaced by noise that is at the same level as the ambient noise (204).

Abstract: A conversion circuit for connecting a two-wire circuit and a four-wire circuit to each other comprises a transformer having a primary side and a secondary side, the primary side being connected to a two-wire circuit having a variable impedance, a first amplifying circuit having a first input terminal connected to the secondary side of the transformer for receiving an audio frequency signal from the two-wire circuit and a second input terminal, the first amplifying circuit having an output terminal connected to a reception-side line of a four-wire circuit, a second amplifying circuit having an input terminal connected to a transmission-side line of the four-wire circuit and an output terminal connected to the secondary side of the transformer for sending out an audio signal on the transmission-side line to the two-wire circuit, a plurality of impedance matching circuits having respective impedances each determined so as to simulate one state of the variable impedance of the two-wire circuit, and a switch for s

Abstract: A key telephone system comprising a terminal device and a main apparatus which exchanges speech signals with the terminal device, the terminal device having a loudspeaking circuit for providing a predetermined amount of loss to either one of first speech signals selected to be transmitted to the main apparatus and second speech signals received from the main apparatus to prevent howling in a loudspeaking conversation mode, said system comprising: a first communication unit provided in the terminal device, for sending the first speech signals to the main apparatus through a first channel and receiving the second speech signals from the main apparatus through a second channel separated from the first channel; second communication unit provided in the main apparatus and connected to the first communication unit, for receiving the first speech signals from the first communication unit through the first channel and sending the second speech signals to the first communication unit through the second channel; in the

Abstract: An active acoustic attenuation system (10) is provided with various adaptive filter models (40, 48, 56, 70, 84, 100) enabling communication between persons (26, 30) in spaced zones (12, 16) by selectively cancelling undesired noise and undesired speech, all on an on-line basis without dedicated off-line pretraining and also for both broadband and narrowband noise.

Abstract: A side tone preventive circuit for telephones which prevents the output of the transmitter on the speaker's side from being transmitted to the receiver on the same side. The balance of the impedance bridge in a bridge circuit for separating a transmission signal and a reception signal is automatically adjusted in accordance with the circuit impedance at the location at which a telephone is installed. An impedance measuring circuit measures the impedance of the telephone circuit, and a balance adjustment circuit the impedance of which is adjustable is provided in the bridge circuit. The impedance of the telephone circuit is measured at the time of installation of a telephone, and the impedance of the balance adjustment circuit is adjusted in correspondence with the result of measurement. Thus, the impedance balance of the bridge circuit is adjusted in correspondence with the circuit impedance, thereby effectively preventing a side tone.

Abstract: A portable device to be carried by a reporter and coupled to a telephone line to a location from which the reporter wishes to transmit a report to a broadcast station to be broadcast live has a casing with a microphone (1) for voice input, a tape input (2) for receiving a tape signal and a monitor (3) for providing acoustic signal to the reporter.The device has signal conditioning circuitry including voice control gain adjustment (20) and amplification (5) for providing a signal to be transmitted by the telephone line in a form suitable for broadcasting. The device is coupled to the telephone line by a hybrid transformer arrangement (7 to 14) which serves to cancel unwanted components of the signal to be sent down a telephone line from that to be received from the telephone line. The device is portable and has its own power supply in the form of batteries or connections for an AC mains auto supply.

Abstract: An office line interface circuit includes a DC circuit for extracting a DC signal component from the office line current, an AC circuit for extracting an AC signal component from the office line current to supply the extracted AC signal component to the receiver of a telephone set, an impedance circuit switchably connected to the office line, and a switching circuit for connecting the impedance circuit to the office line when the extracted DC signal component exceeds a predetermined level to maintain at a substantially constant value the impedance of the office line as viewed from the telephone set. With this interface circuit, the sidetone preventing circuit for the telephone system can operate satisfactorily without complicating the construction under any office line condition.

Abstract: A circuit is disclosed for use in telephone handsets having a line signal node (1), a microphone signal input (20) and an output to an earphone amplifier. There is provided two voltage feedback loops via respective impedance networks (10, 11) and a current feedback loop (5, 7, 14). These feedback loops determine the impedance presented to the line at the signal node with specific voltage/current characteristics and signal frequencies. The first voltage feedback loop in conjunction with the current feedback loop defines the signal frequency impedance and the second voltage feedback loop defines the DC mask presented to the line.

Abstract: A signal processing circuit for a telephone set. This circuit includes signal sending means that sends to a transmission line a sound signal produced from a telephone transmitter, a first adding circuit that adds the signal from the transmission line to the sound signal from the transmitter, an inverting circuit that phase-inverts the output signal from the first adding circuit, an impedance element provided between the transmission line and the output of the inverting circuit, and a second adding circuit that adds the output sound signal from the transmitter to the output signal from the inverting means and supplies the resulting composite signal to a telephone receiver.

Abstract: An amplifier having properties of natural side tone balancing based on the utilization of a bridge arrangement, high efficiency with resulting high gain, high AC impedance and compatibility with electret microphones and electret microphone amplifiers. A matched pair of differential transistors arranged as part of a bridge circuit form a detector of the receive signal at one input versus a transmit signal which appears at both inputs in the same phase and therefore is rejected as a common mode signal. The resultant balancing of side tone and high gain finds particular application in telephone networks where the characteristics dictate the need for higher than usual receive gain or for use by auditorially handicapped users over a normal telephone network.

Abstract: An adaptive hybrid circuit is disclosed for adaptively cancelling a crosstalk component from a transmitter to a receiver in a local station of a local area network. In a preferred embodiment the adaptive hybrid circuit is coupled to a common two-wire transmission line in the network and also between the transmitter and the receiver of the local station. The adaptive hybrid circuit is comprised of a comparator, an integrator and an adjustment circuit. An output signal from the receiver is compared in the comparator with a transmit signal to develop a first signal which causes the integrator to develop a second signal having an amplitude which varies as a function of the states of the first signal. The second signal from the integrator, a transmitter signal, and a composite signal from the transmission line containing a crosstalk component from the transmitter and a received signal component transmitted from a remote station are applied to the adjustment circuit.

Abstract: An adaptive line hybrid for digital telecommunications including a bridge circuit for connection to the telecommunication line the bridge circuit having automatically adjustable components in two arms of the bridge to compensate for variations in the characteristics of the telecommunications line.