Abstract: Provided is a method for audio peak reduction using an all-pass filter, including: determining a delay parameter m and a gain parameter g based on a formula (1): arg ? min m , g max n ? "\[LeftBracketingBar]" y m , g ( n ) ? "\[RightBracketingBar]" , ( 1 ) absolute peak map Y ? ( m , g ) = max n ? "\[LeftBracketingBar]" y m , g ( n ) ? "\[RightBracketingBar]" , ym,g(n) represents a processed signal with a time-domain response function and is calculated based on a formula (2): ym,g(n)=(hs*x)(n)(2), where hs represents an impulse response function, x (n) represents an input signal, and hs is calculated based on formula (3): H S ( z ) = g + z - m 1 + gz - m . ( 3 ) This method is widely used in the reproduction, storage and broadcasting of sound, and the computational complexity is small, which is a supplement to the traditional nonlinear compression algorithm.

Abstract: Provided is a method for audio peak reduction using an all-pass filter, including: determining a delay parameter m and a gain parameter g based on a formula (1): arg ? min m , g max n ? "\[LeftBracketingBar]" y m , g ( n ) ? "\[RightBracketingBar]" , ( 1 ) absolute peak map Y ? ( m , g ) = max n ? "\[LeftBracketingBar]" y m , g ( n ) ? "\[RightBracketingBar]" , ym,g(n) represents a processed signal with a time-domain response function and is calculated based on a formula (2): ym,g(n)=(hs*x)(n)(2), where hs represents an impulse response function, x (n) represents an input signal, and hs is calculated based on formula (3): H S ( z ) = g + z - m 1 + gz - m . ( 3 ) This method is widely used in the reproduction, storage and broadcasting of sound, and the computational complexity is small, which is a supplement to the traditional nonlinear compression algorithm.

Abstract: A method, apparatus and computer program, the method comprising: obtaining information relating to a measured response of a headphone wherein the response is measured when the headphone is positioned adjacent an ear canal of a user; obtaining information relating to a target response of the headphone wherein the target response replicates a different headphone; determining at least one filter using the measured response and the obtained information relating to the target response; and enabling the at least one filter to be applied to an audio signal to obtain the target response.

Abstract: A system for a sports activity, which system is adapted to detect potential game contacts that include contacts between games equipment (1) and a games object (2) and/or contacts between a games object (2) and a target surface (5), and the system comprises sensor means (3a, 3b) adapted to detect vibrations caused by potential game contacts and to convert these vibrations into sensor signals.

Abstract: In a room with strong low-frequency modes the control of excessively long decays is problematic or impossible with conventional passive means. In this patent application a systematic methodology is presented for active modal equalization able to correct the modal decay behaviour of a loudspeaker-room system. Two methods of modal equalization are proposed. The first method modifies the primary sound such that modal decays are controlled. The second method uses separate primary and secondary radiators and controls modal decays with sound fed into at least one secondary radiator. Case studies of the first method of implementation are presented.

Abstract: The present invention relates to a method for designing a modal equalizer (5) for a low frequency audible range, typically for frequencies below 200 Hz for a predetermined space (1) (listening room) and location (2) (location in the room) therein, in which method modes to be equalized are determined at least by center frequency and decay rate of each mode, creating a discrete-time description of the determined modes, and determining equalizer filter coefficients on the basis of the discrete-time description of the determined modes, and forming the equalizer (5) by means of a digital filter by defining the filter coefficients on the basis of the properties of the modes.

Abstract: The invention is concerned with a method and system for modification of audio signals including pitch-bending of the audio signals in accordance with a control signal. The audio signal consists of an input signal defining the point at which the audio signal is received and of a control signal defining the desired change of the signal pitch. The method comprises digitizing and storing samples of the input signal in a data file memory in such a form that each piece of signal data is placed in a queue having an input pointer that locates the next available storage location and output pointer that locates the next location to read from the queue.

Abstract: The present invention relates to a method for designing a modal equalizer (5) for a low frequency audible range, typically for frequencies below 200 Hz for a predetermined space (1) (listening room) and location (2) (location in the room) therein, in which method modes to be equalized are determined at least by center frequency and decay rate of each mode, creating a discrete-time description of the determined modes, and determining equalizer filter coefficients on the basis of the discrete-time description of the determined modes, and forming the equalizer (5) by means of a digital filter by defining the filter coefficients on the basis of the properties of the modes.

Abstract: The invention relates to a method and an equipment for attenuating sound in a duct. Sound propagating in a duct is detected by means of a detector (2) and attenuated by using two successive monopole elements (3, 4) in such a way that both elements function as a dipole approximation and the elements are also used to approximatively produce the monopole radiation needed. A dipole control signal is fed to both elements (3, 4) at a phase shift which is 180° between the two elements. In addition, a monopole control signal is fed to the same elements (3, 4), only this time cophasally. Total volume velocities produced by the two elements (3, 4) are combinations of the portions obtained from the monopole and dipole sources.

Abstract: In a room with strong low-frequency modes the control of excessively long decays is problematic or impossible with conventional passive means. In this patent application a systematic methodology is presented for active modal equalization able to correct the modal decay behaviour of a loudspeaker-room system. Two methods of modal equalization are proposed. The first method modifies the primary sound such that modal decays are controlled. The second method uses separate primary and secondary radiators and controls modal decays with sound fed into at least one secondary radiator. Case studies of the first method of implementation are presented.

Abstract: The invention is related to a method and circuit arrangement for processing a received signal in a variable symbol rate system, such as a digital television system. In the method and arrangement according to the system, a received signal is sampled at a fixed sampling frequency (f.sub.f) that is higher than the symbol frequency of any one of the received signals. The resulting sample sequence is converted to another sample sequence the sampling frequency of which equals the symbol frequency (f.sub.i) of the received signal or its integer multiple. Then the samples are filtered (8) and signal value decisions are made (9) for the filtered samples. Conversion of the sampling frequency is advantageously performed using a so-called modified Farrow-type fractional delay filter (6) which is controlled using a control signal proportional to the delay of each sample.