Abstract: A circuit arrangement for two-wire/four-wire conversion in a DMT system, which is connected to a digital reception path, a digital transmission path and also an analog transmission/reception path and which has nonlinear echo cancellation in the time domain of a signal and also linear echo cancellation in the frequency domain of the signal, furthermore comprises a device for adaptation of the nonlinearities in the frequency domain.

Abstract: Disclosed is an integrated tone detection processor for discriminating between tone and voice signals and determining the tones. The tone detection processor performs Automatic Gain Control (AGC) to normalize the power of the tone or voice signal. Further, the energy of the tone or voice signals are determined at specific frequencies utilizing a Goertzel Filter process. The tone detection processor determines whether or not a tone is present, and if a tone exists, determines the type of tone. Based upon determining the two maximum energy levels of the Goertzel filtered tone, whether the tone is a single tone, dual tone, silence, or other (e.g. speech) can be discriminated. The tone can then be identified by a user-defined dictionary of tones. Based upon various ON and OFF cadence checks in combination with the use of TONE ON and TONE OFF counters, tones can be declared.

Abstract: There is disclosed an echo canceller circuit for use in a full duplex transceiver of the type comprising a line driver capable of sending analog transmit signals through a cable and comprising a line receiver capable of receiving analog receive signals from the cable. An echo canceller impedance model circuit is coupled to an output of the line driver and is coupled to an input of the line receiver. The echo canceller impedance model circuit generates an echo canceller current that is equal in magnitude and opposite in phase to a current that represents signal echoes that are present in the analog receive signals. The echo canceller impedance model circuit has a variable impedance for generating the echo canceller current. The variable impedance has at least one variable resistor and at least one variable capacitor.

Abstract: An enhanced telephone emulation computer system including a minidialer program for controlling a computer to add telephony functions which can be invoked from whatever active program is currently controlling the computer. The minidialer program controls the computer to alter its processing depending upon the context existing at the time when a mouse click or hot key combination event is detected indicating the user wishes to invoke a telephony function. The minidialer program determines whether the user has highlighted any text or numbers in the active window of the application currently controlling the computer and whether the highlighted material is a name or a phone number, and if a name, whether the name is stored with a phone number in a phone book or file maintained on the computer. Processing and telephony menu options displayed as available also depend upon whether the user is or is not on the phone at the time the mouse click or hot key event occurs.

Abstract: A near full duplex portable handset speakerphone comprises: a microprocessor; a hands-free receive register connected to the microprocessor; a hands-free transmit register connected to the microprocessor; a ROM having a speakerphone operation algorithm, the ROM connected to the microprocessor; a first analog-to-digital converter connected to the hands-free receive register; a second analog-to-digital converter connected to the hands-free transmit register; a first programmable digital attenuator connected to the microprocessor and to a speaker; and a second programmable digital attenuator connected to the microprocessor and to a microphone, wherein near full duplex communication is achieved without digital signal processing. In another feature of the invention, the hands-free registers provide a digital representation of the speech volume in each direction to the microprocessor.

Abstract: In a method for adapting a hearing aid, and a hearing aid arrangement for improving the directivity of a hearing aid with a number of microphones that are connected to one another for generating a directional characteristic, the hearing aid is exposed to acoustic waves in a room for precision measuring while the user is wearing it, and the directional characteristic is registered. The filter parameters that arise therefrom are supplied to parameterizable filters in the hearing aid that are connected downstream from the microphones, and the desired ideal directional characteristic can be approximated taking the individual conditions into account when the hearing aid is worn.

Abstract: An observed signal that is corrupted with impulse noise is recorded in a signal processing system of an image processing system or a digital subscriber line (xDSL). The observed signal that is recorded by the signal processing system includes a noise component and data component. The signal processing system estimates the parameters of an alpha-stable distribution using a modified iteratively reweighted least squares (IRLS) technique. The estimated parameters define a probability density function that is used to model the noise component of the observed signal. Once the parameters of the alpha-stable distribution are estimated, the signal processing system uses them to estimate model coefficients of a non-linear prediction filter such as a Volterra filter. Using the model coefficients, the non-linear prediction filter estimates the data component of the observed signal.

Abstract: A telephone apparatus comprises a first audio input device, a second audio input device, a first audio output device, and a second audio output device. The telephone apparatus is operative to couple the first audio input device and the first audio output device to a first telephone line. In response to a first signal, the coupling between the first audio input device and the first telephone line is modified, and the second audio input device and the second audio output device are coupled to a second telephone line.

Abstract: An apparatus and method for remotely and automatically adjusting the volume of a remotely controlled audio device. In one embodiment, the apparatus comprises a sensor circuit for continuously detecting audio signals generated by the audio device, a difference circuit for determining the difference between the amplitude of the detected audio signals and a reference audio signal amplitude and for outputting a signal that represents this difference, a difference signal transfer circuit having an input for receiving the difference signal and an output wherein the difference signal is coupled to the output when the sensor circuit outputs a signal that indicates an audio signal has been detected, and a control circuit for generating a control signal that effects attenuation, augmentation or maintenance of the amplitude of the audio signals generated by the audio device in accordance with the difference signal when the sensor circuit detects an audio signal.

Abstract: Oral instructions for setting up a system of speakers are provided in a setup speaker. The setup speaker audibly plays the instructions and is powered by a battery or external power source. Oral indications of successful connections and completion of the setup are provided. The language of the instructions is also selectable.

Abstract: An echo cancellation unit includes a model store to store a current echo model from an adaptive filter when a real-time error occurs. The real-time error is detected by monitoring a convergence metric. In some embodiments, the convergence metric is echo return loss enhancement (ERLE). When a real-time error occurs, the current echo model is saved, and the adaptive filter is reset such that it will begin converging from the origin. As a new model emerges in the adaptive filter, it is compared to the saved model in the model store at several time lags. If a match is found, the emerging model is replaced with the saved model at the appropriate time lag. The result is faster convergence of the adaptive filter than if the adaptive filter were left to converge on its own.

Abstract: The circuit for adaptive suppression of noise is a component part of a digital-hearing aid, consists of two microphones (1, 2), two AD—converters (3, 4), two compensating filters (5, 6), two retarding elements (7, 8), two subtractors (9, 10), a processing unit (11), a DA—converter (13), an earphone (15) as well as the two filters (17, 18). The method for adaptive suppression of noise can be implemented with the indicated circuit. The two microphones (1, 2), provide two differing electric signals (d1(t), d2(t)), which are digitalized in the two AD—converters (3, 4) and pre-processed together with the two fixed compensation filters (5, 6). Downstream the compensation filters are arranged the two filters (17, 18) symmetrically crosswise in a forward direction and having adaptive filter coefficients (w1, w2). The filter coefficients (w1, w2) are calculated by a stochastic gradient procedure and updated in real time while minimizing a quadratic cost function consisting of cross-correlation terms.

Abstract: In a telecommunications network short dialling codes for a subscriber terminal are programmed by another party, and the allocation of codes is communicated to the user. An audio announcement played as part of the dial tone may be used to communication the allocation of codes.

Abstract: Disclosed is a method and apparatus which facilitates the use of existing power that is ordinarily unused or discarded to provide power to a telephone interface circuit when the circuit is in the on-hook state, thereby alleviating the need to use loop current to power the interface during the on-hook state. Existing clock signals are used to charge capacitors that are normally used for capacitive coupling of digital data across the high voltage isolation barrier. Although only small-value capacitors are needed for transmitting data across the capacitive interface, larger-value capacitors are used and are configured to form a charge pump to generate power to the interface at all times. Thus the interface always has a steady source of power available for use, including during the on-hook state for powering circuitry that can detect, modulate, and transmit on-hook signals across the capacitive interface.

Abstract: A method for providing echo cancellation between a first and second communication unit detects an echo condition (200) between first and second communication units. In addition, the method includes detecting whether a decoded near end audio packet contains a suitable level of background noise to be used as a background noise packet. If so, the current near end encoded audio packet is stored (202) in a buffer (26), such as a circular buffer. Previous near end encoded audio packets (42) that are also determined to have a suitable level of background noise based on their decoded corresponding counterparts are also stored in the circular buffer. When an echo condition is detected with respect to a current near end encoded audio packet, the method includes substituting a previous near end encoded audio packet stored in the buffer as the next encoded audio packet for transmission (204).

Abstract: An adaptive noise canceling microphone system for extracting a desired signal, in particular a desired speech signal, comprising two microphones being arranged at a predefined distance from each other; a signal forming system (SFS) being adapted to receive a first and second input signals resulting from sounds received by the two microphones wherein an acoustical signal component in the first input signal is determined, wherein an acoustical signal component in the second input signal is determined, wherein the acoustical signal component in the first input signal is enhanced to generate a speech enhanced signal, and wherein the acoustical signal component in the second input signal is suppressed to generate a speech nulled signal; an adaptive noise cancellation filtering circuit being adapted to receive the speech enhanced signal and the speech nulled signal, wherein the noise in the speech enhanced signal is cancelled using the speech nulled signal as reference, thereby generating an output filtered signal

Abstract: An apparatus for disabling a telephone ringer includes a light sensing device operable to produce a signal indicative of a level of ambient light; and a controller operable to receive the signal and disable the telephone ringer when the signal indicates that the ambient light has reached a predetermined level.

Abstract: A DFT is used to compute the magnitude and phase of the frequencies desired to be detected in a frame of sampled data by performing 2 dot products for each such frequency. The two dot products are calculated using the samples in the frame and an equal number of equally spaced cosine and sine coefficients for each frequency to be detected.

Abstract: An active noise cancellation system has a speaker which is utilized to generate sound to cancel engine noise, as is known. An operator horn switch communicates with a control for the active noise cancellation system. Upon receiving a request from the operator for a horn signal, the control for the active noise cancellation system generates a horn tone from the speaker associated with the active noise cancellation system. If cancellation is in progress, the cancellation is preferably paused during the period of time when the horn signal is being emitted.

Abstract: A circuit and related method for digital volume control are provided, where the circuit includes a digital filter configured to process samples of an input stream in a manner that processes a previous input sample during a time interval before a subsequent input sample, and outputs a series of exponentially decaying waveforms. The result is an exponential response to a volume change made by a user, where the change feels more pleasant and natural than a conventional linear response.