A signal processing method comprises comparing a first frequency domain representation of a sequence of power values for respective windows of source input samples of a source input signal with a second frequency domain representation of a sequence of power values for respective windows of target input samples of a target input signal so as to generate a frequency domain difference representation; inverse-frequency-transforming the frequency domain difference representation to generate a modification indication; and applying the modification indication to the source input samples to generate respective output samples of an output signal.

A method for speech equalization, comprising the steps of receiving an input audio signal, processing said input audio signal in dependence on frequency and to providing an equalized electric audio signal according to an equalization function, wherein said equalization function comprises at least an actuator part configured to dynamically applying a compensation filter to the received input signal and dynamically applying a transparent filter to the received input signal, and further transmitting an output signal perceivable by a user as sound representative of said electric acoustic input signal or a processed version thereof.

The present disclosure relates to an audio system and a method of controlling the same. An aspect of the present disclosure is to provide an optimal listening state according to a user setting volume and hardware characteristics of an audio system. The audio system includes a signal inputter configured to receive an input audio signal; a signal processor configure to variably apply a dynamic range for each area of the input audio signal, to apply a scaling of a size corresponding to an inverse number of a boosting gain determined by a user setting volume for the input audio signal, to perform a tone mapping of the input audio signal by applying the boosting gain, to perform signal compensation for the input audio signal by the inverse number of the scaling, and to apply a master volume according to the user setting volume; a storage configured to store parameters used in signal processing of the signal processor; and a signal outputter configured to output the audio signal processed by the signal processor.

A hearing device comprises an input transducer providing an input gain G.sub.I, a signal processor comprising a compressor for determining a frequency and level dependent desired compressor gain G.sub.P to compensate for a hearing impairment of the user, and to provide a resulting compressor gain G′.sub.P, and an output transducer for providing output stimuli perceivable as sound for the user based on a processed signal, the output transducer providing an output gain, G.sub.O. A resulting forward path gain G′ is defined in a logarithmic representation as G.sub.I+G′.sub.P+G.sub.O. The hearing device further comprises a loop gain estimator for continuously estimating a current loop gain ΔL(n), configured to provide a loop gain estimate within a predefined number of feedback loop delays after a feedback buildup has started, and a loop gain controller for dynamically controlling said resulting forward path gain G′ in dependence of said estimate of said current loop gain ΔL(n). A resulting loop gain, LG′, is determined as a sum of the resulting forward path gain G′ and a feedback gain H when given in a logarithmic representation. The loop gain controller is configured to provide that the resulting loop gain is limited to stay below a predefined value.

A method for controlling the distortion generated by a system having at least one loudspeaker on board a vehicle, the loudspeaker being designed to receive an audio signal. The method includes measuring at least one indicator of the distortion of the at least one loudspeaker, determining an acceptable distortion threshold for each distortion indicator, which can be used to determine a maximum acceptable amplitude for each frequency in a frequency range of interest of the audio signal entering the loudspeaker system, generating an assembly comprising at least one filter, correcting the audio signal entering the loudspeaker system by applying at least one filter determined in the generating step.

A method of improving sound quality of an ultrasonic speaker device includes: a first operation of removing a low frequency component of an input audio signal by using a high pass filter; a second operation of correcting a response characteristic of a signal obtained by removing the low frequency component, for each frequency band by using an equalizer; a third operation of adjusting a sound pressure level of a signal obtained by correcting the response characteristic for each frequency band; and a fourth operation of controlling a signal magnitude not to increase above a particular level by using a limiter, in order to prevent a signal clipping that may occur in the second operation and the third operation. An audio signal of improved sound quality may be transmitted to a listener by applying a sound quality improving method suitable for the characteristics of a super-directional ultrasonic speaker.

An audio device with an electro-acoustic transducer and a processor that is configured to determine if input audio signals are speech-based, and if the input audio signals are determined to be speech-based apply speech dynamic range compression to the input audio signals, to develop revised audio signals. The revised audio signals are provided to the transducer.

The present application relates to a method, system, and computer program product of dynamically adjusting thresholds of a compressor responsive to an input audio signal. A scene switch analyzer receives an input audio signal having a plurality of frequency band components. The scene switch analyzer determines whether a scene switch has occurred in the input audio signal. The frequency band components of the input audio signal are processed. In response to determine that scene switch has not occurred, a distortion audibility system applies slow smoothing to compressor thresholds of the frequency band components. In response to determine that scene switch has occurred, the distortion audibility system applies fast smoothing or no smoothing to the compressor thresholds of the frequency band components.

The present application relates to a method, system, and computer program product of dynamically adjusting thresholds of a compressor responsive to an input audio signal. A scene switch analyzer receives an input audio signal having a plurality of frequency band components. The scene switch analyzer determines whether a scene switch has occurred in the input audio signal. The frequency band components of the input audio signal are processed. In response to determine that scene switch has not occurred, a distortion audibility system applies slow smoothing to compressor thresholds of the frequency band components. In response to determine that scene switch has occurred, the distortion audibility system applies fast smoothing or no smoothing to the compressor thresholds of the frequency band components.

A processor-based method of reducing loudspeaker-induced harmonic distortion in sound generated by a loudspeaker responsive to a first audio signal, the method including: attenuating a first band in the first audio signal by an attenuation gain, the first band including a fundamental frequency, wherein the attenuating dynamically adjusts the attenuation gain in accordance with, at least, a ratio between: i) an energy of the first band in the first audio signal; and ii) an energy of a second band in the first audio signal, the second band including one or more frequencies that is an integer multiple of the fundamental frequency.