Abstract: The present invention relates to a cancellation of echoes in telecommunications systems, more specifically it relates to adaptive alignment of a linear filter (500) used for echo cancellation. According to the invention, it is continuously determined, by means of control logic (520), if a reflection replica delay included in an echo replica signal (110), which delay is provided by a signal buffer (510), should be attempted to be increased or not. Similarly, it is continuously determined if the reflection replica delay should be attempted to be decreased or not. In this way it is possible to provide a delay of the reflection replica which corresponds to the pure delay of a corresponding reflection included in an echo signal (120) received over an echo path. The invention is advantageous since the filter (500) will continuously and quickly adapt to changes in the echo path delay by continuously increasing or decreasing, in an incremental and smooth manner, a present replica delay.

Abstract: A method for mixing data streams (BS) is specified, whereby the mixing is carried out in a digital signal processor (DSP) of an ISDN interface module (IC), and the control of the data streams (BC) is managed by a module for communication according to the Media Gateway Control Protocol (MGCP), which replaces the control of a B channel in the ISDN interface module (IC). Furthermore, a telecommunication terminal with an ISDN interface module (IC) is specified for implementing the method according to the invention.

Abstract: The invention relates to a method for transmitting proprietary signalings between a terminal (1) and a switching center (5, 8) in a telecommunications network (6) for controlling features, whereby the terminal (1) is connected to a first switching center (7), and the proprietary signalings are transmitted between the terminal (1) and a second switching center (5, 8) via a first user data channel (11). The invention also relates to a terminal (1), a switching center (5) and to a communications system (13) for implementing said method.

Abstract: The invention relates to the operation and administration of a call center, the network elements of which or the applications of which are distributed in a network and, in particular, are available via the Internet. In this arrangement, switches of a public network are linked into the administration. Communication between different network elements, particularly applications and servers from different domains takes place by converting interfaces of the applications into a protocol which is intended for interdomain communication.

Abstract: Methods, systems, and arrangements enable true wide-band calibration of analog-to-digital conversion (ADC) using correction table indexing. The frequency of an analog input signal that corresponds to a digital output sample is accounted for, at least partially, when accessing a correction table during the creation and use thereof. For example, in addition to at least a portion of a current sample, at least portion(s) of previous and/or subsequent samples may be used to build (e.g., by bit concatenation) an index for addressing a correction table memory. In effect, compensation may be achieved for ADC errors that are frequency dependent. This correction table indexing may advantageously be employed along with a scheme that estimates one or more parameters of the analog input signal in the digital domain on a sample-by-sample basis in order to reconstruct the analog input signal in the digital domain.

Abstract: Noisy speech parameters are enhanced by determining a background noise power spectral density (PSD) estimate, determining noisy speech parameters, determining a noisy speech PSD estimate from the speech parameters, subtracting a background noise PSD estimate from the noisy speech PSD estimate, and estimating enhanced speech parameters from the enhanced speech PSD estimate.

Abstract: A method, arrangement, and system enable adaptive calibration of an analog-to-digital converter (ADC) from reference signals with unknown parameter(s) that may vary (e.g., a varying frequency). An analog reference signal s(t) is supplied to an ADC to be calibrated. The output x(n) of the ADC is used by a, e.g., sine-wave reconstruction filter to reconstruct an estimate ŝ(n) of the sampled instances of the signal s(t). A recursive frequency estimator within the sine-wave reconstruction filter produces a frequency estimate of the analog reference signal s(t) from the signal x(n). An adaptive reconstruction filter uses this frequency estimate and the signal x(n) to produce the estimate ŝ(n). When a convergence detector determines that the adaptive reconstruction filter has converged, the estimate ŝ(n) is used to alter values that are stored in a correction table. During functional operation of the ADC, the correction table is used to correct the output values of the ADC.

Abstract: A method and system enables ADCs to be calibrated from reference signals with unknown parameters. A given analog reference signal s(t) is supplied to an ADC. The output (x(k)) of the ADC is used by calibration logic to estimate at least one parameter of the reference signal. A FIR filter accepts as input the x(k) signals and outputs an estimate (s(k)) as the sampled instances of the s(t) signal. A reconstruction table is created that approximates the analog input signal in the digital domain using the knowledge of the analog input signal waveform type. The actual ADC outputs are compared to the values in the reconstruction table to produce a correction table for calibration. A continuous time reference signal may be utilized for training the correction table. The method and system is data adaptive, so it may be applied, for example, with any sinusoidal reference input.

Abstract: A residual error based compensator for the frequency domain bias of an autoregressive spectral estimator is disclosed. LPC analysis is performed on the residual signal and a parametric PSD estimate is formed with the obtained LPC parameters. The PSD estimate of the residual signal multiplies the PSD estimate of the input signal.

Abstract: A spectral subtraction noise suppression method in a frame based digital communication system is described. Each frame includes a predetermined number N of audio samples, thereby giving each frame N degrees of freedom. The method is performed by a spectral subtraction function H(w) which is based on an estimate of the power spectral density of background noise of non-speech frames and an estimate .PHI..sub.x (w) of the power spectral density of speech frames. Each speech frame is approximated by a parametric model that reduces the number of degrees of freedom to less than N. The estimate .PHI..sub.x (w) of the power spectral density of each speech frame is estimated from the approximative parametric model.

Abstract: An electric wheel-drive for motor vehicles, particularly automobiles with internal combustion engine, which can be installed nondestructively-reversibly in place of the brakes, with brushless, polyphasic, electronically speed-controlled electric motors individually attached to the wheels. The electric motor is a homopolar multiple-airgap axial-field motor whose rotor (1) replaces the wheeldrum and brake-disk, and whose stator (2) replaces the brake-shoes, brake-pads, and brake splash-shield plate of at least two wheels of a pair, in a conventional automobile with internal combustion engine. An electronic control system is included for the control of the axial-field motor during propulsion and braking. This electric wheel-drive allows for a fast, uncomplicated, and inexpensive hybridization of the car without modifications of the wheels, or the axle, or of other parts of the car, thereby providing the car with second, independent, propulsion system.