Abstract: A tripod for video-photographic equipment, includes: at least one leg having a first, a second and a third section that engage so as to axially slide in a telescopic manner one after the other; a first clamping device that is provided between the first and the second section and a second clamping device that is provided between the second and the third section, an actuating element that is connected to the first clamping device, a transmission element that connects the first and the second clamping device such that the movement of the first clamping device causes the second clamping device to move in a similar manner so that said first and second clamping devices simultaneously clamp and release, and an adjustment member that is provided at the first clamping device in order to adjust the second clamping device via the transmission element.

Abstract: The present disclosure relates to an audio device (900) for providing an improved three-dimensional sound experience by means of the generated soundfield. To achieve this, the audio device (900) comprises a housing (901), which has an elliptical torus shape and a plurality of loudspeakers (903a-903h), and a processing circuitry (1310). The processing circuitry is configured to process a plurality of input signals (L, R, UL, UR) in a manner, which enables the plurality of loudspeakers (903a-903h) to form at least a first (DH1, DH3) and second (DH2) horizontal dipoles for crosstalk cancellation within at least two different frequency ranges (HF, MF), and to form at least a first vertical dipole (DV1, DV3) for sound elevation (1204a, 1204b) of the soundfield. Hereby, the desired frequency ranges (HF, MF) may be adjusted using an appropriated distance of the plurality of loudspeakers (903a-903h).

Abstract: A tripod for video-photographic equipment, includes: at least one leg having a first, a second and a third section that engage so as to axially slide in a telescopic manner one after the other; a first clamping device that is provided between the first and the second section and a second clamping device that is provided between the second and the third section, an actuating element that is connected to the first clamping device, a transmission element that connects the first and the second clamping device such that the movement of the first clamping device causes the second clamping device to move in a similar manner so that said first and second clamping devices simultaneously clamp and release, and an adjustment member that is provided at the first clamping device in order to adjust the second clamping device via the transmission element.

Abstract: The disclosure relates to a microphone assembly (100) for acquiring a plurality of audio signals, wherein the microphone assembly (100) has a reconfigurable geometry so that the microphone assembly (100) may be configured to be embedded in or attached to a body. The microphone assembly (100) comprises: a plurality of digital microphones (101a-h) configured to convert the sound signal impinging on each digital microphone into a corresponding digital electrical signal, a digital signal processing unit (102) comprising a serial digital communication interface (102a) and a processor (102b), and a connecting and mounting structure configured to provide a flexible electrical connection and a flexible mechanical arrangement for the plurality of digital microphones (101a-h).

Abstract: The disclosure relates to an audio signal processing apparatus for processing an input audio signal, comprising a filter unit comprising a plurality of filters, each filter configured to filter the input audio signal to obtain a plurality of filtered audio signals, each filter designed according to an extended mode matching beamforming applied to a surface of a half revolution, the surface partially characterizing a loudspeaker enclosure shape, a plurality of scaling units, each scaling unit configured to scale the plurality of filtered audio signals using a plurality of gain coefficients to obtain a plurality of scaled filtered audio signals, and a plurality of adders, each adder configured to combine the plurality of scaled filtered audio signals, thereby providing an output audio signal for producing a sound field having a beam directivity pattern defined by the plurality of gain coefficients.

Abstract: The present disclosure relates to a method and to an apparatus, both arranged for controlling acoustic signals to be recorded or reproduced by an electro-acoustical sound system. An initial digital filter is determined by solving an inverse problem, wherein the initial digital filter is configured to control acoustic signals to be recorded and/or reproduced by the electro-acoustical sound system; a frequency-dependent articulation parameter is determined by executing a time spectral psychoacoustic automatic audio quality test on the initial digital filter; a frequency-dependent regularization parameter, used for determining the initial digital filter, is tuned by use of the frequency-dependent articulation parameter; and, by use of the tuned frequency-dependent regularization parameter, a digital filter configured to control acoustic signals to be recorded or reproduced by the electro-acoustical sound system is determined.

Abstract: The disclosure relates to a microphone assembly (100) for acquiring a plurality of audio signals, wherein the microphone assembly (100) has a reconfigurable geometry so that the microphone assembly (100) may be configured to be embedded in or attached to a body. The microphone assembly (100) comprises: a plurality of digital microphones (101a-h) configured to convert the sound signal impinging on each digital microphone into a corresponding digital electrical signal, a digital signal processing unit (102) comprising a serial digital communication interface (102a) and a processor (102b), and a connecting and mounting structure configured to provide a flexible electrical connection and a flexible mechanical arrangement for the plurality of digital microphones (101a-h).

Abstract: The disclosure relates to an audio signal processing apparatus for processing an input audio signal, comprising a filter unit comprising a plurality of filters, each filter configured to filter the input audio signal to obtain a plurality of filtered audio signals, each filter designed according to an extended mode matching beamforming applied to a surface of a half revolution, the surface partially characterizing a loudspeaker enclosure shape, a plurality of scaling units, each scaling unit configured to scale the plurality of filtered audio signals using a plurality of gain coefficients to obtain a plurality of scaled filtered audio signals, and a plurality of adders, each adder configured to combine the plurality of scaled filtered audio signals, thereby providing an output audio signal for producing a sound field having a beam directivity pattern defined by the plurality of gain coefficients.

Abstract: A sound field device is disclosed that comprises an elevation cue estimator, a low-frequency filter estimator, and a high-frequency filter estimator. The elevation cue-estimator is configured to estimate an elevation cue of a head-related transfer function (HRTF) of at least one listener. The low-frequency filter estimator is configured to estimate one or more low-frequency filter elements based on the elevation cue. The high-frequency filter estimator is configured to estimate one or more high-frequency filter elements based on the elevation cue. An estimation method of the low-frequency filter estimator is different from an estimation method of the high-frequency filter estimator. The one or more low-frequency filter elements and the one or more high-frequency filter elements are for driving an array of loudspeakers to generate an elevated sound impression at a bright zone.

Abstract: An audio signal processing apparatus for processing an input audio signal is provided, the apparatus comprising a plurality of filters, each filter configured to filter the input audio signal to obtain a plurality of filtered audio signals, each filter designed according to an extended mode matching beamforming applied to a surface of a half revolution, the surface partially characterizing a loudspeaker enclosure shape, a plurality of scaling units, each scaling unit configured to scale the plurality of filtered audio signals using a plurality of gain coefficients to obtain a plurality of scaled filtered audio signals, and a plurality of adders, each adder configured to combine the plurality of scaled filtered audio signals, thereby providing an output audio signal for producing a sound field having a beam directivity pattern defined by the plurality of gain coefficients.

Abstract: The present invention relates to a method and to an apparatus, both arranged for controlling acoustic signals to be recorded or reproduced by an electro-acoustical sound system. For this purpose: an initial digital filter is determined by solving an inverse problem, wherein the initial digital filter is configured to control acoustic signals to be recorded and/or reproduced by the electro-acoustical sound system; a frequency-dependent articulation parameter is determined by executing a time spectral psychoacoustic automatic audio quality test on the initial digital filter; a frequency-dependent regularization parameter, used for determining the initial digital filter, is tuned by use of the frequency-dependent articulation parameter; and, by use of the tuned frequency-dependent regularization parameter, a digital filter configured to control acoustic signals to be recorded or reproduced by the electro-acoustical sound system is determined.

Abstract: The disclosure relates to an apparatus for generating a sound field on the basis of an input audio signal. The apparatus comprises a plurality of transducers, wherein each transducer is configured to be driven by a transducer driving signal ql of the respective transducer; a plurality of filters configured to generate for each transducer the transducer driving signal ql of the respective transducer; and a control unit configured to provide or receive a first transducer driving signal vector q0 of dimension L such that the gradient of J(q;?) with respect to q is zero in (q0;?0), the control unit is further configured to provide a second transducer driving signal vector {tilde over (q)} of dimension L such that the gradient of the cost function J(q;?) with respect to q is [approximately] zero in ({tilde over (q)}; {tilde over (?)}), the control unit is configured to provide the second transducer driving signal vector {tilde over (q)}.

Abstract: The disclosure relates to an apparatus for generating a sound field on the basis of an input audio signal. The apparatus comprises a plurality of transducers, wherein each transducer is configured to be driven by a transducer driving signal ql of the respective transducer; a plurality of filters configured to generate for each transducer the transducer driving signal ql of the respective transducer; and a control unit configured to provide or receive a first transducer driving signal vector q0 of dimension L such that the gradient of J(q;?) with respect to q is zero in (q0;?0), the control unit is further configured to provide a second transducer driving signal vector {tilde over (q)} of dimension L such that the gradient of the cost function J(q;?) with respect to q is [approximately] zero in ({tilde over (q)};{tilde over (?)}), the control unit is configured to provide the second transducer driving signal vector {tilde over (q)}.

Abstract: The disclosure relates to an audio signal processing apparatus for processing an input audio signal, comprising a filter unit comprising a plurality of filters, each filter configured to filter the input audio signal to obtain a plurality of filtered audio signals, each filter designed according to an extended mode matching beamforming applied to a surface of a half revolution, the surface partially characterizing a loudspeaker enclosure shape, a plurality of scaling units, each scaling unit configured to scale the plurality of filtered audio signals using a plurality of gain coefficients to obtain a plurality of scaled filtered audio signals, and a plurality of adders, each adder configured to combine the plurality of scaled filtered audio signals, thereby providing an output audio signal for producing a sound field having a beam directivity pattern defined by the plurality of gain coefficients.

Abstract: A sound field device is disclosed that comprises an elevation cue estimator, a low-frequency filter estimator, and a high-frequency filter estimator. The elevation cue-estimator is configured to estimate an elevation cue of a head-related transfer function (HRTF) of at least one listener. The low-frequency filter estimator is configured to estimate one or more low-frequency filter elements based on the elevation cue. The high-frequency filter estimator is configured to estimate one or more high-frequency filter elements based on the elevation cue. An estimation method of the low-frequency filter estimator is different from an estimation method of the high-frequency filter estimator. The one or more low-frequency filter elements and the one or more high-frequency filter elements are for driving an array of loudspeakers to generate an elevated sound impression at a bright zone.

Abstract: An apparatus for estimating an overall mixing time, where the apparatus comprises a processing element configured to determine differences between energy profiles of a first room impulse response of the first pair of room impulse responses and a second room impulse response of the first pair of room impulse responses at a plurality of different sample times of the first pair of room impulse responses, set a sample time of the plurality of sample times as a mixing time for the first pair of room impulse responses at which the difference between the energy profiles of the first room impulse response and the second room impulse response of the first pair of room impulse responses is equal to or below a threshold value, and determine the overall mixing time based on the mixing time for the first pair of room impulse responses.

Abstract: An apparatus for estimating an overall mixing time, where the apparatus comprises a processing element configured to determine differences between energy profiles of a first room impulse response of the first pair of room impulse responses and a second room impulse response of the first pair of room impulse responses at a plurality of different sample times of the first pair of room impulse responses, set a sample time of the plurality of sample times as a mixing time for the first pair of room impulse responses at which the difference between the energy profiles of the first room impulse response and the second room impulse response of the first pair of room impulse responses is equal to or below a threshold value, and determine the overall mixing time based on the mixing time for the first pair of room impulse responses.

Abstract: A leveling device for laying tiles or the like includes a tie-rod having a support base for edge portions of one or more adjacent tiles, a projecting element from the base, which can be arranged into junctions between adjacent tiles, and a knob coupling with the projecting element and having an outlet pass-through hole for a free end of the projecting element. The knob is configured to force an edge portion of the adjacent tiles against the base to level the placement and is configured to receive and retain, by snap coupling elements, a blocking tie-rod hooking and retaining the projecting element inside a pass-through channel defined in the blocking tie-rod. The blocking tie-rod is sectioned into two coupling half-shells, embracing and constraining the projecting element when inserted in the cap, and mutually separating to leave free the projecting element when at least partially extracted from the cap.

Abstract: An apparatus and a method for compressing a set of N binaural room impulse responses, BRIR, wherein each channel of an N channel audio signal is convolved with the corresponding compressed set of N BRIR. The apparatus may comprise at least one analyzing and compressor module adapted to separate an input binaural room impulse response signal into a first binaural signal set provided to the binauralization processing of the initial part of the BRIR (early part) and a second binaural signal set provided to the binauralization processing of the final part of the BRIR (late part) via a downmix module; a binauralization module adapted to obtain a binaural signal based on convolving the N channel audio signal with the first binaural signal set and the second binaural signal set.

Abstract: A leveling device for laying tiles or the like includes a tie-rod having a support base for edge portions of one or more adjacent tiles, a projecting element from the base, which can be arranged into junctions between adjacent tiles, and a knob coupling with the projecting element and having an outlet pass-through hole for a free end of the projecting element. The knob is configured to force an edge portion of the adjacent tiles against the base to level the placement and is configured to receive and retain, by snap coupling elements, a blocking tie-rod hooking and retaining the projecting element inside a pass-through channel defined in the blocking tie-rod. The blocking tie-rod is sectioned into two coupling half-shells, embracing and constraining the projecting element when inserted in the cap, and mutually separating to leave free the projecting element when at least partially extracted from the cap.