3274results about "Frequency response correction" patented technology

A speaker includes at least one transducer and at least one memory device. The at least one transducer is adapted to receive an audio signal. The at least one memory device is adapted to store data related to the speaker. A method includes reading data from a memory device of at least one speaker. An audio signal is provided from an audio system.

A prefilter (5) for an audio system comprising a loudspeaker (1) in a room (2), which corrects both amplitude and phase errors due to the loudspeaker (1) by a linear phase correction filter response and corrects the amplitude response of the room (2) whilst introducing the minimum possible amount of extra phase distortion by employing a minimum phase correction filter stage. A test signal generator (8) generates a signal comprising a periodic frequency sweep with a greater phase repetition period than the frequency repetition period. A microphone (7) positioned at various points in the room (2) measures the audio signal processed by the room (2) and loudspeaker (1), and a coefficient calculator (6) (e.g. a digital signal processor device) derives the signal response of the room and thereby a requisite minimum phase correction to be cascaded with the linear phase correction already calculated for the loudspeaker (1). Filter (5) may comprise the same digital signal processor as the coefficient calculator (6). Applications in high fidelity audio reproduction, and in car stereo reproduction.

The invention relates to a method and a system for adapting a loudspeaker to a specific listening position relative to the loudspeaker according to which method and system the acoustic power radiated by the loudspeaker is corrected by means of a correction filter inserted in the signal path through the loudspeaker, the response of said correction filter being determined by comparison between the a quantity characterising the radiated acoustic power measured at an actual listening position and a similar quantity measured at a reference listening position. According to a specific embodiment of the invention said characterising quantities are the radiation resistances measured at the actual listening position and the reference listening position respectively.

Control system for devices such as an audio reproduction system, an actuator device, an electromechanical device and a telephony device. The system includes control circuitry which receives an input signal and a signal indicative of a position of a portion of the controlled apparatus. The control circuit provides an output signal to the controlled apparatus to affect an operation of the controlled apparatus. The output signal provides control of the apparatus to compensate for one or more of: motor factor; spring factor; back electromotive force; and impedance of a coil in a driver of the controlled apparatus. The signal indicative of position is derived by one or more position indicator techniques such as an infrared LED and PIN diode combination, position dependent capacitance of one portion of the controlled apparatus with respect to another portion of the controlled apparatus, and impedance of a coil in the controlled apparatus. The control circuitry is configurable to control transconductance and / or transduction of the system being controlled. A technique is disclosed to detect and measure a cant of a voice coil transducer, the technique including measuring a capacitance between one portion of the voice coil transducer with respect to another portion of the voice coil transducer over a range of movement of the voice coil during operation.

In one embodiment, a directional microphone array having (at least) two microphones generates forward and backward cardioid signals from two (e.g., omnidirectional) microphone signals. An adaptation factor is applied to the backward cardioid signal, and the resulting adjusted backward cardioid signal is subtracted from the forward cardioid signal to generate a (first-order) output audio signal corresponding to a beampattern having no nulls for negative values of the adaptation factor. After low-pass filtering, spatial noise suppression can be applied to the output audio signal. Microphone arrays having one (or more) additional microphones can be designed to generate second- (or higher-) order output audio signals.

A method, apparatus and system are disclosed herein for loudspeaker equalization. In one embodiment, the system comprises an input for receiving samples of an input signal, a pre-compensator to produce a pre-compensated output in response to the samples of an input signal, parameters of a loudspeaker model, and previously predistorted samples of the input signal, and a loudspeaker, corresponding to the loudspeaker model, to produce an audio output in response to the pre-compensated output.

Techniques and systems for enhancing delivered audio signals are disclosed which may be employed in a delivery system at a server side, a client side, or both. The techniques include forming a processed audio signal by processing audio signals through multiple pathways which operate on different frequency bands using dynamic processing and other elements, and thereafter providing recording or listening environment enhancements and other sound enhancements to the processed audio signal. Also disclosed are techniques and systems for implementing the multi-pathway processing and environmental and sound enhancements.

Improved vocal signals are obtained in headsets and similar devices by including a microphone inside a chamber formed at least in part by the wearer's ear. This second microphone provides a reduced noise input signal. The reduced noise signal is corrected by input from another microphone, located outside the chamber. This correction can include echo cancellation, spectral shaping, frequency extension, and the like.

Loudspeakers can be damaged by high drive signals. One reason for this damage is an excess vibration displacement of the coil-diaphragm assembly. This invention describes a novel method for limiting this displacement by a signal processor. In the present invention, a low frequency shelving and notch filter is used to attenuate low frequencies according to a prediction of the loudspeaker displacement. A novel method for calculating coefficient values for a digital implementation of the low frequency shelving and notch filter according to the predicted displacement is described.

A headset having game, chat and microphone audio signals is provided with a programmable signal processor for individually modifying the audio signals and a memory configured to store a plurality of user-selectable signal-processing parameter settings that determine the manner in which the audio signals will be altered by the signal processor. The parameter settings collectively form a preset, and one or more user-operable controls can select and activate a preset from the plurality of presets stored in memory. The parameters stored in the selected preset can be loaded into the signal processor such that the sound characteristics of the audio paths are modified in accordance with the parameter settings in the selected preset.

Control system for devices such as an audio reproduction system, an actuator device, an electromechanical device and a telephony device. The system includes control circuitry which receives an input signal and a signal indicative of a position of a portion of the controlled apparatus. The control circuit provides an output signal to the controlled apparatus to affect an operation of the controlled apparatus. The output signal provides control of the apparatus to compensate for one or more of: motor factor; spring factor; back electromotive force; and impedance of a coil in a driver of the controlled apparatus. The signal indicative of position is derived by one or more position indicator techniques such as an infrared LED and PIN diode combination, position dependent capacitance of one portion of the controlled apparatus with respect to another portion of the controlled apparatus, and impedance of a coil in the controlled apparatus. The control circuitry is configurable to control transconductance and / or transduction of the system being controlled. A technique is disclosed to detect and measure a cant of a voice coil transducer, the technique including measuring a capacitance between one portion of the voice coil transducer with respect to another portion of the voice coil transducer over a range of movement of the voice coil during operation.

A noise reduction circuit for reducing the effects of low frequency noise such as wind noise in communications applications is described. In one embodiment, the noise reduction circuit features a high pass filter formed by exploiting the existing off-chip AC coupling capacitances in making the connection to the source of audio signals. The filter may be adaptive to environmental low frequency noise level through programming the shunt resistances. A low-noise wide dynamic range programmable gain amplifier is also described. Adaptive equalization of the audio signal is also described through the utilization of programmable front-end resistors and a back-end audio equalizer.

The loudspeaker device according to the present invention comprises a loudspeaker; a feedforward processing section for performing feedforward processing on an electric signal to be inputted to the loudspeaker based on a preset filter coefficient so that non-linear distortion which occurs from the loudspeaker is removed; and a feedback processing section for detecting vibration of the loudspeaker, and performing feedback processing on an electric signal concerning the vibration with respect to the electric signal to be inputted to the loudspeaker. The feedback processing section performs feedback processing on the electric signal concerning the vibration so that the non-linear distortion which occurs from the loudspeaker is removed and so that a frequency characteristic concerning the vibration of the loudspeaker becomes a predetermined frequency characteristic.

A method and apparatus for detecting microphone interference includes first and second built-in microphones producing first and second microphone signals. A first filter bank creates first high-frequency-band and first low-frequency-band signals from the first microphone signal. A second filter bank creates second high-frequency-band and second low-frequency-band signals from the second microphone signal. A first measurement calculator determines a high-frequency-band energy value from the first high-frequency-band signal and the second high-frequency-band signal when the first and second high-frequency-band signals' magnitudes exceeds predetermined thresholds. A second measurement calculator calculates a low-frequency-band energy value from the first low-frequency-band signal and the second low-frequency-band signal when the first and second low-frequency-band signals' magnitudes exceed predetermined thresholds. A logic control block, coupled to the first measurement calculator and the second measurement calculator, detects microphone interference and produces an output signal indicating microphone occlusion or wind noise.

An automatic sound field correcting device executes a signal process to the plurality of audio signals on respective correspondent signal transmission paths, and outputs them to a plurality of correspondent speakers to correct sound characteristics on the respective signal transmission paths. Namely, a measurement signal is supplied to each signal transmission path, and a measurement sound corresponding to it is outputted from the speaker to a sound space. The outputted measurement sound is detected as a detecting signal. The frequency characteristic of the audio signal on each signal transmission path is corrected by an equalizer, and a gain value of the equalizer is determined by a correction amount determining unit. A frequency characteristics correction is performed predetermined times. At a first correction, the correction amount determining unit determines the correction amount by performing a frequency analysis, based on the detecting signal, i.e. base on the detecting signal corresponding to the measurement sound actually outputted to the sound space. On the contrary, at and after a second correction, the correction amount determining unit determines the correction amount based on the detecting signal or an output signal of the equalizer. Namely, at and after the second correction, the output signal of the equalizer is supplied to the correction amount determining unit in a signal processing circuit as the need arises, and the frequency characteristics correction is performed without actually outputting the measurement sound to the sound space.

An arrangement and method for converting an input signal into a mechanical or acoustical output signal by using a transducer and additional means for generating a desired transfer behavior and for protecting said transducer against overload. Transducers of this kind are for example loudspeaker, headphones and other mechanical or acoustical actuators. The additional means comprise a controller, a power amplifier and a detector. The detector identifies parameters of the transducer model if the stimulus provides sufficient excitation of the transducer. The detector permanently identifies time variant properties of the transducer for any stimulus supplied to the transducer. The controller provided with this information generates a desired linear or nonlinear transfer behavior; in particular electric control linearizes, stabilizes and protects the transducer against electric, thermal and mechanical overload at high amplitudes of the input signal.

The invention relates to an arrangement and a method for converting an input signal v into an output signal p(ra) by using an electro-mechanical transducer and for reducing nonlinear total distortion pd in said output signal p(ra), whereas the nonlinear total distortion pd contains multi-modal distortion ud which are generated by nonlinear partial vibration of mechanical transducer components. An identification system generates distributed parameters Pd of a nonlinear wave model (Nd) and lumped parameters Pl of a network model (Nl) based on electrical, mechanical or acoustical state variables of transducer measured by a sensor. The nonlinear wave model distinguishes between activation modes and transfer modes, whereas the activation modes affect the transfer modes, which transfer the input signal u into the output signal p. A control system synthesizes based on the physical modeling and identified parameters Pd and Pl nonlinear distortion signals vd and vl which are supplied with the input signal v to the transducer and compensate for the distortion signals ul and ud generated by the transducer nonlinearities.

A speaker system for a musical instrument that detects the displacement of a voice coil of a speaker and provides feedback processing. The speaker system has a preamp that alters the frequency characteristics of the an electrical signal that has been input to an input terminal, and a power amplifier that amplifies the electrical signal. A speaker is driven by the power amplifier and a feedback unit detects the displacement of the speaker and provides a feedback signal to the power amplifier. The power amplifier amplifies the electrical signal in conformance with the output of the preamp and the feedback signal.

A method of compensating for spatial audio frequency characteristics that varies in accordance with a mounting condition of a down firing speaker of an audio / video (AV) apparatus includes calculating a listening distance between the AV apparatus and a listener, calculating a distance between a speaker mounted on the AV apparatus and a neighboring reflective surface, setting a spatial frequency compensation filter value and a speaker frequency characteristic compensation filter value based on the calculated distances, and compensating for frequency characteristics of an audio signal by combining the spatial frequency compensation filter value and the speaker frequency characteristic compensation filter value.

A method and apparatus in a sound-reproduction system, in which method an electrical calibration signal is formed, an audio signal is formed in the loudspeaker from the calibration signal, the response of the audio signal is measured and analysed, and the loudspeaker system is adjusted on the basis of the measurement results. The operator is permitted to made additional alterations to the settings of the loudspeaker system on the basis of the measurement performed, the effects of the alterations are calculated and displayed to the operator without additional measurements, and the additional settings are implemented in real time in the loudspeaker system.

Control system for devices such as an audio reproduction system, an actuator device, an electromechanical device and a telephony device. The system includes control circuitry which receives an input signal and a signal indicative of a position of a portion of the controlled apparatus. The control circuit provides an output signal to the controlled apparatus to affect an operation of the controlled apparatus. The output signal provides control of the apparatus to compensate for one or more of: motor factor; spring factor; back electromotive force; and impedance of a coil in a driver of the controlled apparatus. The signal indicative of position is derived by one or more position indicator techniques such as an infrared LED and PIN diode combination, position dependent capacitance of one portion of the controlled apparatus with respect to another portion of the controlled apparatus, and impedance of a coil in the controlled apparatus. The control circuitry is configurable to control transconductance and / or transduction of the system being controlled. A technique is disclosed to detect and measure a cant of a voice coil transducer, the technique including measuring a capacitance between one portion of the voice coil transducer with respect to another portion of the voice coil transducer over a range of movement of the voice coil during operation. COMPUTER PROGRAM LISTING APPENDIX

Embodiments disclosed herein enable detection and improvement of the quality of the audio signal using a mobile device by determining the loss in the audio signal and enhancing audio by streaming the remainder portion of audio. Embodiments disclosed herein enable an improvement in the sound quality rendered by rendering devices by emitting an test audio signal from the source device, measuring the test audio signal using microphones, detecting variation in the frequency response, loudness and timing characteristics using impulse responses and correcting for them. Embodiments disclosed herein also compensate for the noise in the acoustic space by determining the reverberation and ambient noise levels and their frequency characteristics and changing the digital filters and volumes of the source signal to compensate for the varying noise levels.

A method is proposed for automatically adjusting the filter parameters—center frequency, quality and amplification or attenuation—of at least one digital equalizer which is a component of a reproduction device for audio signals in a vehicle passenger compartment. To that end, first of all, the acoustical frequency response of the passenger compartment is ascertained. The inadequacies in the acoustics of the passenger compartment in the form of local maxima and minima in the measured frequency response are then determined. On this basis, the filter parameters are adjusted automatically so that at least a portion of these inadequacies is compensated. A reproduction device for audio signals for implementing this method is also proposed.

Control system for devices such as an audio reproduction system, an actuator device, an electromechanical device and a telephony device. The system includes control circuitry which receives an input signal and a signal indicative of a position of a portion of the controlled apparatus. The control circuit provides an output signal to the controlled apparatus to affect an operation of the controlled apparatus. The output signal provides control of the apparatus to compensate for one or more of: motor factor; spring factor; back electromotive force; and impedance of a coil in a driver of the controlled apparatus. The signal indicative of position is derived by one or more position indicator techniques such as an infrared LED and PIN diode combination, position dependent capacitance of one portion of the controlled apparatus with respect to another portion of the controlled apparatus, and impedance of a coil in the controlled apparatus. The control circuitry is configurable to control transconductance and / or transduction of the system being controlled. A technique is disclosed to detect and measure a cant of a voice coil transducer, the technique including measuring a capacitance between one portion of the voice coil transducer with respect to another portion of the voice coil transducer over a range of movement of the voice coil during operation.

The sound device includes an audio-information output unit, an analysis unit, an audio-division-spectrum output unit, a noise-division-spectrum output unit and a correction unit. The analysis unit receives audio information from the audio-information output unit, and then outputs sound spectrum information. The noise-division-spectrum output unit outputs sound-volume information for each critical band width of a noise, and the audio-division-spectrum output unit outputs the sound-volume information for each critical band width of the sound-spectrum information. The correction unit corrects the information from the audio-division-spectrum output unit based on the information from the noise-division-spectrum output unit. The audio-signal properties can be well corrected corresponding to the auditory-sense properties of the human, and thus the audio sound, in which an uncomfortable feeling to the auditory sense of the human has been adequately controlled, can be transmitted to a user.

A method and an apparatus for compensating for nonlinear distortion are provided to divide audio signals reproduced in a nonlinear speaker system into linear and nonlinear components in a time domain and a frequency domain, and then generate inversely-corrected signals by means of an inverse filtering scheme, so that it is possible to further consider a variety of nonlinear distortion characteristics such as viscous damping and structural damping which have not been reflected in the conventional lumped parameter method, and thus to obtain better sound quality.

An active noise cancellation system reduces, at a listening position, the power of a noise signal being radiated from a noise source to the listening position. The system includes an adaptive filter that receives a reference signal representing the noise signal, and provides a compensation signal. A bass management unit receives the compensation signal and applies a phase shift to the compensation signal to provide a phase shifted compensation signal. A first acoustic radiator receives the phase shifted compensation signal and radiates audio indicative thereof to the listening position. A second acoustic radiator receives the compensation signal and radiates audio indicative thereof to the listening position. The transfer function characteristics from the input of the bass management system to the listening position approximately matches a desired transfer function.

The present publication describes a calibration method and apparatus, in which an electrical calibration signal is formed, an audio signal is formed in the loudspeaker from the calibration signal, the response of the audio signal is measured and analysed, and the system is adjusted on the basis of the measurement results. The calibration signal is formed in the loudspeaker in such a way that it is essentially a sinusoidal signal, the frequency of which scans at least substantially through the entire audio frequency range.