Abstract: The invention relates to an arrangement for improving leak tolerance in an earpiece (100) of a radio device. The invention can be applied preferably in mobile stations. One idea of the invention is that an acoustic volume (111) is arranged behind the earpiece by taking advantage of the RF shield casing (112, 116) of a radio frequency unit of the device. By means of the invention one can form behind the earpiece an acoustic volume which loads optimally the earpiece without therefore needing to enlarge the size of the device. In an embodiment of the invention one wall of the casing arranged behind the earpiece has been formed by means of an electronic circuit board (112), to which circuit board radio frequency components (117) of an electronic unit have been connected. Then between the casing and the external volume one can preferably form an acoustic path by means of through holes (113, 114) arranged in the circuit board.

Abstract: The present invention provides a microphone array including a small number of real microphone that can realize the same characteristics as a microphone array including a large number of real microphones. The microphone array of the present invention includes a plurality of real microphones, at least one virtual microphone, and an estimator for estimating a sound signal to be received by the virtual microphone based on the sound signals received by the real microphones.

Abstract: An echo canceler for suppressing an echo of a receive signal contained in a transmit signal directed to a telephone circuit and transferring the transmit signal subjected to echo suppression to the telephone circuit. The echo canceler includes a pseudo echo path for attenuating and delaying the receive signal supplied from the telephone circuit and adding the delayed receive signal to the transmit signal, and an echo cancel circuit for estimating a transfer function of the pseudo echo path on the basis of the receive signal supplied from the telephone circuit and the echo of the receive signal, generating a pseudo echo on the basis of a value of the estimation and the receive signal, subtracting the pseudo echo from the transmit signal passed through the pseudo echo path, and thereby transferring the transmit signal subjected to echo suppression to the telephone circuit.

Abstract: A feedback controller, embedded in an FM audio processor, that controls the “integrated multiplex power” to the requirements of ITU-R 412 (2.51) is disclosed. This regulation specifies the maximum power, which is the same as the power produced by a sinewave modulating the carrier ±19 kHz (where ±75 kHz is 100% peak modulation). The controller applies the square of the multiplex signal to an integrator. A constant is removed from the integrator; representing the maximum power threshold. The output of the integrator is sampled periodically with a sample-and-hold circuit. A second integrator receives the output of the sample-and-hold. The output of the second integrator circuit is added to the threshold setting of a compressor, which determines the average power output of the compressor. The output of the compressor is applied through a peak controller to the input of the multiplex coder, closing a feedback loop, which controls the integrated multiplex power to a preset threshold.

Abstract: A method for improving acoustic sidetone suppression in hands-free telephones, in particular for use in motor vehicles, is described. The telephone has a level discriminator and a plurality of adaptive echo compensation filters each processing a subband. In at least one subband, a further adaptive filter (a shadow filter) of a lower order is connected in parallel with the adaptive echo compensation filter. Spatial changes are detected with the aid of a combined evaluation of a correlation analysis and a residual error comparison of the two competing, adaptive filters.

Abstract: An audio interface for coupling a telephone handset and a telephone base unit to a computer is provided. The audio interface comprises a feed through circuit coupled between the telephone handset and the telephone base unit and a buffer circuit connected to the computer. The buffer circuit is coupled to the feed through circuit, in one example, by one or more transformers that perform a bi-directional conversion between balanced signals of the feed through circuit and single-ended signals of the buffer circuit. In one example, the feed through circuit comprises a microphone circuit and an earpiece circuit. In one example, the buffer circuit has an output of a first buffer connected to an audio input of the computer and an input of a second buffer connected to an audio output of the computer.

Abstract: An echo canceller circuit for use in an echo canceller system is set forth that provides sensitive double-talk detection. The echo canceller circuit comprises a second digital filter having adaptive tap coefficients to stimulate an echo response occurring during the call. The adaptive tap coefficients of the second digital filter are updated over the duration of the call using a Least Mean Squares process having an adaptive gain a. A channel condition detector is used to detect channel conditions during the call. The channel condition detector is responsive to detected channel conditions for changing the adaptive gain a during the call. For example, the channel condition detector may detect the presence of a double-talk condition and set the adaptive gain a to zero. Similarly, the channel condition detector may detect the occurrence of a high background noise condition and set the adaptive gain a to a level less than 1 that is dependent on the detected level of the background noise.

Abstract: A incoming-call tone generation device comprises: a first storage unit 1, for storing an effect calculation formula for incoming-call tone data; a second storage unit, for storing incoming-call tone data that consist of data for original sounds; a controller, for reading incoming-call tone data from the second storage unit and transmitting the data to a digital signal processor 4, and for controlling a D/A converter 5; the digital signal processor 4, for calculating an effect pattern and a sine wave using the calculation formula read from the first storage unit, and for performing an envelope process for the incoming-call tone data; the D/A converter 5, for converting, into analog data, digital incoming-call tone data obtained during the envelope process, and for outputting an analog incoming-call tone; and a loudspeaker 6 through which an analog incoming-call tone is released.

Abstract: The invention is a method and apparatus for performing echo cancellation utilizing a Proportionate Normalized Least Mean Squares (PNLMS) algorithm having a high convergence rate and low complexity. The invention utilizes a proportional step-size adaptive algorithm which utilizes the signs of the regressor rather than the regressor value itself in the gradient calculation.

Abstract: Noise is removed from the digitized output of a sensor, subject to undesired resonance, even when the resonant frequency is unknown or drifts, with sufficiently low phase delay for the sensor to be used in closed-loop control. A very narrow notch filter which removes the resonance-induced noise is recursive (IIR) and therefore has a low phase delay. However, the apparatus which determines the center frequency of the notch filter is non-recursive, and therefore stable. It includes a tunable FIR filter which tracks the same resonance that we wish the IIR filter to remove. Tuning the FIR filter to minimize the output of the FIR filter therefore tunes the notch frequency to align with the resonant frequency. The tuning parameter which adaptively produces this result is suitably scaled and biased, and is applied to the IIR filter.

Abstract: A surge protection device has a modular construction so that a manufacturer can assemble the parts in any combination desired by a customer. This eliminates any need to maintain large varieties of surge protection modules in inventory. The modular construction includes a housing having a plurality of bores formed therein into which is press fit long pins and short pins for electrical communication with an external power source and an internal customer telephone circuit, respectively. The housing further includes a pair of recesses into each of which may be press fit a positive temperature coefficient resistor. A first embodiment of spring clips helps hold the positive coefficient resistors in place and provides needed electrical contacts, and a second embodiment of spring clips further includes an elongate part that extends to a test point opening formed in a cover for the modular device.

Abstract: A load management system for monitoring the output from a power supply and managing the load on said power supply, said load management system comprising a hysteresis circuit for signaling a load element to stop drawing current when an output voltage at said output falls below a first threshold and for signaling said load element to resume drawing current when said output voltage rises above a second threshold, said hysteresis circuit comprising a first state sensing circuit having an input coupled to said output and for signaling whether said output voltage is greater than or less than said first voltage threshold, a second state sensing circuit having an input coupled to said output and for signaling whether said output voltage is greater than or less than a second voltage threshold, and a state transition circuit having a first input coupled to the output of said first state sensing circuit and a second input coupled to the output of said second state sensing circuit, said state transition circuit having a

Abstract: A method of canceling echo signals in a telephone network operating in a double talk mode has been developed. A system for implementing the method is also presented. In the invention the conventional normalized least mean square (NLMS) algorithm for echo cancellers is modified such that echo path changes continue to be tracked even after a double talk condition has been detected. The modified algorithm, known herein as a double-talk normalized least mean square (DNLMS) algorithm adaptively adjusts the convergence rate based on the power difference between the echo signal or the residual signal and the far end signal.

Abstract: A method of anticipating an access code such as a telephone number. A database of reference access codes is provided. As each character of a new access code is dialed, the database is searched for reference access codes that have the same leading digits. If one match is found, that matching reference access code is displayed to allow the user to call it without dialing new characters. If more than one match is found, the matching reference access codes are ranked according to a selection criterion. Preferably, the database includes an indication of the history of usage of the reference access codes and the selection criterion is based on the usage histories of the reference access codes, for example, according to their times of most recent usage, and also according to other criteria such as current day of the week. The highest ranking reference code is displayed. If the user completes dialing with no match found, then the newly dialed access code is included in the database.

Abstract: A microphone bias current detection circuit includes: a microphone circuit 18; an amplifier 10 having a first output and a second output, the first output is coupled to the microphone circuit 18 for providing a bias current to the microphone circuit 18, the second output provides a sampled current Is proportional to the bias current; a first switch 30 having a first end coupled to the second output of the amplifier 10; a resistor 38 having a first end coupled to a second end of the first switch 30; and a second switch 32 coupled between the first end of the resistor 38 and a reference current source.

Abstract: A system and method for canceling an echo signal. An input waveform is provided to an acoustic processor, and a determination is made whether the input waveform includes information representative of an echo signal. If the input waveform includes information representative of an echo signal, an output waveform is formed by attenuating a residual waveform with the acoustic processor. The residual waveform is attenuated by an attenuation factor that gradually changes from an initial attenuation value to a final attenuation value during the attenuation step. A system and method for adjusting an acoustic signal from a muted state to an unmuted state by varying an attenuation factor applied to an acoustic signal by an acoustic processor. The acoustic signal is provided to an acoustic processor. An output waveform is formed from the acoustic processor by adjusting the attenuation factor from a muted state to a first attenuation value associated with the non-muted state.

Abstract: A processor controlled mechanism automatically determines from a location such as a central office whether termination equipment installed at a remote end of a wireline link is span-powered, or is powered locally. It then applies a voltage to the wireline link that is appropriate for the identified type of termination equipment. This eliminates the need for two separate central office loop-powering devices. In the course of making the determination, nothing higher than a safety extra low voltage is applied to the wireline link, to comply with Underwriter's Laboratory standards for locally powered customer premises equipment.

Abstract: A system and method of the present invention conceal errors caused by lost audio in an audio transmission. A frame error detector detects audio data lost in an audio data transmission. An audio decoder generates frequency and time domain data from received audio data. A transient detector detects the presence of a transient audio signal in the received audio data. A frame synthesizer interpolates frequency domain data to generate synthetic audio data to construct audio data in place of the lost audio data.

Abstract: A telephone system incorporating the present invention provides the capability for automated dialing of a telephone number stored in a caller ID log by determining the correct dialing procedure and then dialing the telephone number in accordance with the correct dialing procedure.

Abstract: An echo cancellation device (ECD) comprises an echo canceller (EC) including a transfer function estimator (EST, H) and a subtractor (ADD) and a residual echo suppression device (G, ADD2). The residual echo suppression device (G) comprises a residual echo filter (G) having an adjustable filter function (g). This filter function (g) can be adapted to either remove from the subtractor output (TNE′) the spectral characteristics relating to the reception signal (RFE) and/or to emphasize in the subtractor output signal (TNE′) a background signal spectral content relating to the transmission signal (TNE). A noise generation means (NGM′) can be provided at the output of the adaptable filter (G) for injecting a noise process in to the filter output signal (TNE′) prior to a speech coding in a speech coder (COD). The noise process masks in the filter output signal a spectral content relating to the reception signal (RFE).