Fraunhofer-gesellschaft Zur Foerderung Der Angewandten Forschung E.v.

Andreas Franck

Christoph Sladeczek

Thomas Sporer

Sandra Brix

Michael Rath

Walter Kellermann

Daniel Beer

Martin Schneider

Albert Zhykhar

The effectiveness of calculating FIR filter coefficients for beam-forming filters for transducer arrays such as arrays of microphones or loudspeakers, for example, is increased in that the calculation is performed in two stages; namely, on the one hand, by calculating frequency domain filter weights of the beam-forming filters, i.e., coefficients describing the transfer functions of the beam-forming filters within the dimension of the frequency so as to obtain target frequency responses for the beam-forming filters, so that applying the beam-forming filters to the array approximates a desired directional selectivity, and followed by calculating the FIR filter coefficients for the beam-forming filters, i.e., of coefficients describing the impulse response of the beam-forming filters within the time domain, such that the frequency responses of the FIR beam-forming filters approximate the target frequency responses in an optimum manner in accordance with defined criteria.

FIR filter coefficient calculation for beam-forming filters

A device for calculating loudspeaker signals for a plurality of loudspeakers while using a plurality of audio sources, an audio source including an audio signal, includes a forward transform stage for transforming each audio signal, block-by-block, to a spectral domain so as to obtain for each audio signal a plurality of temporally consecutive short-term spectra, a memory for storing a plurality of temporally consecutive short-term spectra for each audio signal, a memory access controller for accessing a specific short-term spectrum among the plurality of short-term spectra for a combination consisting of a loudspeaker and an audio signal on the basis of a delay value, a filter stage for filtering the specific short-term spectrum for the combination of the audio signal and the loudspeaker by using a filter provided for the combination of the audio signal and the loudspeaker, so that a filtered shot-term spectrum is obtained for each combination of an audio signal and a loudspeaker, a summing stage for summing up the filtered short-term spectra for a loudspeaker so as to obtain summed-up short-term spectra for each loudspeaker, and a backtransform stage for backtransforming, block-by-block, summed-up short-term spectra for the loudspeakers to a time domain so as to obtain the loudspeaker signals.

Device and method for calculating loudspeaker signals for a plurality of loudspeakers while using a delay in the frequency domain

A device for processing a signal includes a processor stage configured to filter the signal present in a frequency-domain representation by a filter with a filter characteristic in order to obtain a filtered signal, to provide the filtered or a signal derived from the filtered signal with a frequency-domain window function, in order to obtain a windowed signal, wherein providing has multiplications of frequency-domain window coefficients of the frequency-domain window function by spectral values of the filtered signal or the signal derived from the filtered signal in order to obtain multiplication results, and summing up the multiplication results. Further, the device has a converter for converting the windowed signal or a signal determined using the windowed signal to a time domain in order to obtain the processed signal.

Device and method for processing a signal in the frequency domain

A device for processing a signal includes a processor stage configured to filter the signal present in a frequency-domain representation by a filter with a filter characteristic in order to obtain a filtered signal, to provide the filtered or a signal derived from the filtered signal with a frequency-domain window function, in order to obtain a windowed signed, wherein providing has multiplications of frequency-domain window coefficients of the frequency-domain window function by spectral values of the filtered signal or the signal derived from the filtered signal in order to obtain multiplication results, and summing up the multiplication results. Further, the device has a converter for converting the windowed signal or a signal determined using the windowed signal to a time domain in order to obtain the processed signal.

A loudspeaker system for a vehicle includes a loudspeaker array including a plurality of electroacoustic sound transducers that can be controlled individually, such that a user-specific audio signal can be reproduced for different users at different listening positions in a vehicle interior of the vehicle via the plurality of electroacoustic sound transducers. Here, the loudspeaker array or a sound outlet of the loudspeaker array is arranged in particular between at least two of the listening positions in the vehicle interior, i.e. for example between the driver and the passenger seat.

Loudspeaker system

A more clear-cut separation of a first audio signal within a first region of a sonication area of a plurality of loudspeakers is achieved in that a calculator calculates that version of the audio signals which results from the spatially selective reproduction of the audio signals at this first region, in that a masking threshold is calculated as a function of the version of that audio signal which is to be separated from the one or more other audio signals at this region, and in that the emission of the audio signals for spatially selective reproduction to the outputs of the plurality of loudspeakers is influenced as a function of a comparison of the masking threshold with the version of the one or more other, i.e. spurious, audio signals.

Device and method for spatially selective audio reproduction

A device for generating a multitude of loudspeaker signals based on a virtual source object which has a source signal and a meta information determining a position or type of the virtual source object. The device has a modifier configured to time-varyingly modify the meta information. In addition, the device has a renderer configured to transfer the virtual source object and the modified meta information to form a multitude of loudspeaker signals.

Device and method for decorrelating loudspeaker signals

A device for calculating loudspeaker signals using a plurality of audio sources, an audio source including an audio signal, includes a forward transform stage for transforming each audio signal to a spectral domain to obtain a plurality of temporally consecutive short-term spectra, a memory for storing a plurality of temporally consecutive short-term spectra for each audio signal, a memory access controller for accessing a specific short-term spectrum for a combination consisting of a loudspeaker and an audio signal based on a delay value, a filter stage for filtering the specific short-term spectrum by using a filter, so that a filtered short-term spectrum is obtained for each audio signal and loudspeaker combination, a summing stage for summing up the filtered short-term spectra for a loudspeaker to obtain summed-up short-term spectra, and a backtransform stage for backtransforming summed-up short-term spectra for the loudspeakers to a time domain to obtain the loudspeaker signals.

A device for determining a component signal for a WFS system includes a provider for providing WFS parameters, a WFS parameter interpolator, and an audio signal processor. The provider provides WFS parameters for a component signal while using a source position and while using the loudspeaker position at a parameter sampling frequency smaller than the audio sampling frequency. The WFS parameter interpolator interpolates the WFS parameters so as to produce interpolated WFS parameters which are present at a parameter interpolation frequency that is higher than the parameter sampling frequency, the interpolated WFS parameters having interpolated fractions which have a higher level of accuracy than is specified by the audio sampling frequency. The audio signal processor is configured to apply the interpolated fractional values to the audio signal such that the component signal is obtained in a state of having been processed at the higher level of accuracy.

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Ilmenau, Germany

Andreas Franck