Abstract: In a method for determining the flow rate of fluids using the ultrasonic transit time method, the flow speed and therefrom the flow rate of the fluid is determined from the transit time of ultrasound signals through the measurement section. In order to take into account the change in the transmission function of the ultrasonic converters, which change acts as a drift of the measurement variable, the transmission function of the measurement section is determined at least approximately, and at least one ultrasonic transit time is corrected by means of a correction value that is determined from the group transit time of the transmission function.

Abstract: In a method for determining the flow rate of fluids using the ultrasonic transit time method, the flow speed and therefrom the flow rate of the fluid is determined from the transit time of ultrasound signals through the measurement section. In order to take into account the change in the transmission function of the ultrasonic converters, which change acts as a drift of the measurement variable, the transmission function of the measurement section is determined at least approximately, and at least one ultrasonic transit time is corrected by means of a correction value that is determined from the group transit time of the transmission function.

Abstract: A sensor system is located in an environment composed of a first medium, where waves propagate with a first phase velocity, the sensor system including at least one main enclosure and a sensor array with at least two sensors, said sensor array being arranged inside the main enclosure, wherein the space inside the main enclosure between the sensor array and the inner surface of the main enclosure is filled with a second medium, in which waves propagate with a second phase velocity, the second phase velocity being different from the first velocity.

Abstract: The invention relates to a method for localizing and tracking acoustic sources (101) in a multi-source environment, comprising the steps of recording audio-signals (103) of at least one acoustic source (101) with at least two recording means (104, 105), creating a two- or multi-channel recording signal, partitioning said recording signal into frames of predefined length (N), calculating for each frame a cross-correlation function as a function of discrete time-lag values (?) for channel pairs (106, 107) of the recording signal, evaluating the cross-correlation function by calculating a sampling function depending on a pitch parameter (f0) and at least one spatial parameter (?0), the sampling function assigning a value to every point of a multidimensional space being spanned by the pitch-parameter and the spatial parameters, and identifying peaks in said multidimensional space with respective acoustic sources in the multi-source environment.

Abstract: In order to improve the speech quality of an electric larynx (EL) speaker, the speech signal of which is digitized by suitable means, the following steps are carried out: a) dividing a single-channel speech signal into a series of frequency channels by transferring it from a time domain into a discrete frequence domain; b) filtering out the modulation frequency of the EL by way of a high-pass or notch filter, in each frequency channel; and c) back-transforming the filtered speech signal from the frequency domain into the time domain and combining it into a single-channel output signal.

Abstract: A sensor system being is located in an environment composed of a first medium, where waves propagate with a first phase velocity, the sensor system including at least one main enclosure and a sensor array with at least two sensors, said sensor array being arranged inside the main enclosure, wherein the space inside the main enclosure between the sensor array and the inner surface of the main enclosure is filled with a second medium, in which waves propagate with a second phase velocity, the second phase velocity being different from the first velocity.

Abstract: A method for creating a form of a non-linear filter suitable for reducing a computational complexity is proposed. The filter is resolved into polyphase components in such a way that the polyphase components can be interchanged with a conversion of the sampling rate of a signal to be sent to the filter or of a signal to be emitted by the filter. Corresponding filters and filter arrangements are also proposed. In this way, a computational complexity for calculating the signal to be emitted by the filter can be significantly simplified. The invention can be used in echo compensation.

Abstract: The invention relates to a method for localizing and tracking acoustic sources (101) in a multi-source environment, comprising the steps of recording audio-signals (103) of at least one acoustic source (101) with at least two recording means (104, 105), creating a two- or multi-channel recording signal, partitioning said recording signal into frames of predefined length (N), calculating for each frame a cross-correlation function as a function of discrete time-lag values (?) for channel pairs (106, 107) of the recording signal, evaluating the cross-correlation function by calculating a sampling function depending on a pitch parameter (f0) and at least one spatial parameter (?0), the sampling function assigning a value to every point of a multidimensional space being spanned by the pitch-parameter and the spatial parameters, and identifying peaks in said multidimensional space with respective acoustic sources in the multi-source environment.

Abstract: An apparatus for noise suppression having a linear prediction analysis circuit having an LP error filter (LFF), which takes a first, noisy voice signal y(n)=x(n)+?(n) as a basis for producing an LP-error-filter output signal e(n), having a coefficient calculation unit (KBE), which updates the coefficients of the LP error filter on the basis of the internal signals (including the input and out signals y(n) and e(n)) in the LP error filter, and having a subtraction unit, which subtracts the LP error filter output signal e(n) from the first voice signal y(n) in a subtractor and, following the subtraction, outputs the remainder as a second voice signal x(n)=y(n)?e(n) in which the noise is suppressed.

Abstract: An electrical network having a nonlinear transfer response is approximated with a system with memory. The system with memory being approximated in the frequency domain and subsequently being expanded in the time domain. A transfer response of the system being approximated to a transfer response of the electrical network in a range of a system bandwidth corresponding to an input signal bandwidth. The resulting model has adjustable parameters and can readily be implemented in the form of a dynamically linear filter and a static nonlinearity (B2) connected thereto.

Abstract: A model network of a nonlinear circuitry includes one or more static nonlinear elements and a plurality of linear filters with transfer functions. A method for determining the model network includes performing an input amplitude-to-output amplitude measurement of the nonlinear circuitry and performing an input amplitude-to-output phase measurement of the nonlinear circuitry. The transfer functions are calculated on the basis of results of the input amplitude-to-output amplitude measurement and input amplitude-to-output phase measurement.

Abstract: An arrangement for a time interleaved analog-to-digital converter that converts an signal to a digital signal and has a converter array with a plurality of analog-to-digital converters arranged in a fixed sequence in parallel with one another and can be operated with staggered timing with respect to one another is disclosed. The arrangement has a connection network which, for the purposes of actuation with staggered timing, generates in each case one control signal for an individual analog-to-digital converter in each case, with the connection network predefining the time sequence with which the control signals actuate the individual analog-to-digital converters in such a way that owing to this sequence of the control signals and thus the sequence of the actuated individual analog-to-digital converters there is at least a reduction in an interference spectrum in the spectrum of the input and/or output signal. A sorting method for operating this analog-to-digital converter is also disclosed.

Abstract: A model network of a nonlinear circuitry includes one or more static nonlinear elements and a plurality of linear filters with transfer functions. A method for determining the model network includes performing an input amplitude-to-output amplitude measurement of the nonlinear circuitry and performing an input amplitude-to-output phase measurement of the nonlinear circuitry. The transfer functions are calculated on the basis of results of the input amplitude-to-output amplitude measurement and input amplitude-to-output phase measurement.

Abstract: An electrical network having a nonlinear transfer response is approximated with a system with memory. The system with memory being approximated in the frequency domain and subsequently being expanded in the time domain. A transfer response of the system being approximated to a transfer response of the electrical network in a range of a system bandwidth corresponding to an input signal bandwidth. The resulting model has adjustable parameters and can readily be implemented in the form of a dynamically linear filter and a static nonlinearity (B2) connected thereto.

Abstract: An arrangement for a time interleaved analog-to-digital converter that converts an signal to a digital signal and has a converter array with a plurality of analog-to-digital converters arranged in a fixed sequence in parallel with one another and can be operated with staggered timing with respect to one another is disclosed. The arrangement has a connection network which, for the purposes of actuation with staggered timing, generates in each case one control signal for an individual analog-to-digital converter in each case, with the connection network predefining the time sequence with which the control signals actuate the individual analog-to-digital converters in such a way that owing to this sequence of the control signals and thus the sequence of the actuated individual analog-to-digital converters there is at least a reduction in an interference spectrum in the spectrum of the input and/or output signal. A sorting method for operating this analog-to-digital converter is also disclosed.

Abstract: A method for creating a form of a non-linear filter suitable for reducing a computational complexity is proposed. The filter is resolved into polyphase components in such a way that the polyphase components can be interchanged with a conversion of the sampling rate of a signal to be sent to the filter or of a signal to be emitted by the filter. Corresponding filters and filter arrangements are also proposed. In this way, a computational complexity for calculating the signal to be emitted by the filter can be significantly simplified. The invention can be used in echo compensation.