A method for adjusting the clarity of an audio output in a changing environment, including: receiving a content signal; applying a customized gain to the content signal; and outputting the content signal with the customized gain to at least one speaker for transduction to an acoustic signal, wherein the customized gain is applied on a per frequency bin basis such that frequencies of a lesser magnitude are enhanced with respect to frequencies of a greater magnitude and an intelligibility of the acoustic signal is set approximately at a desired level, wherein the customized gain is determined according to at least one of a gain applied to the content signal, a bandwidth of the content signal, and a content type encoded by the content signal.

A method of generating audio loudness performed by an audio loudness generation device may include: receiving loudness information on each of a plurality of audio tracks included in one group; predicting an intermediate loudness distribution, which is a loudness distribution for the one group, on the basis of the loudness information on each of the plurality of audio tracks; and generating an integrated loudness for the one group on the basis of the intermediate loudness distribution.

In some embodiments, a method for performing enhancement on an audio signal to generate an enhanced audio signal in response to feedback indicative of amount of compression applied to at least one frequency band of the enhanced audio signal. In typical embodiments, the enhancement is or includes bass enhancement. Examples of other types of enhancement performed in other embodiments include dialog enhancement, upmixing, frequency shifting, harmonic injection or transposition, subharmonic injection, virtualization, and equalization. Other aspects are systems (e.g., programmed processors) and devices (e.g., devices having physically-limited bass reproduction capabilities, such as, for example, a notebook, tablet, mobile phone, or other device with small speakers) configured to perform any embodiment of the method.

The invention relates to a hearing aid a cochlear implant comprising a) at least one input transducer for capturing incoming sound and for generating electric audio signals which represent frequency bands of the incoming sound, b) a sound processor which is configured to analyze and to process the electric audio signals, c) a transmitter that sends the processed electric audio signals, d) a receiver/stimulator, which receives the processed electric audio signals from the transmitter and converts the processed electric audio signals into electric pulses, e) an electrode array embedded in the cochlear comprising a number of electrodes for stimulating the cochlear nerve with said electric pulses, and f) a control unit configured to control the distribution of said electric pulses to the number of said electrodes. The control unit is configured to distribute said electric pulses to the number of said electrodes by applying one out of a plurality of different coding schemes, and wherein the applied coding scheme is selected according to characteristics of the incoming sound.

The present disclosure provides a method, an apparatus and a device for processing a sound signal, wherein the method comprises: acquiring a transmitted signal spectrum of a target sound signal sent out by a loudspeaker and a received signal spectrum of the target sound signal received by a microphone; detecting whether there is a signal distortion frequency band with signal distortion in the target sound signal according to the transmitted signal spectrum and the received signal spectrum, and when detecting that the signal distortion frequency band exists, performing compression processing on the target sound signal according to the signal distortion frequency band during a current signal processing cycle, and transmitting a compressed target sound signal through the loudspeaker.

A system and method for optimizing the low-frequency sound rendition of an audio signal, implementing variations in a plurality of parameters of the audio signal according to the volume level of the signal chosen by a user, in particular filtering or compression parameters, or parameters relating to the harmonics of the audio signal, while seeking to optimize the dynamics and the bandwidth of the audio signal according to the volume, in order to provide an optimal rendition to the user.

Methods, systems, and computer program products of automatic de-essing are disclosed. An automatic de-esser can be used without manually setting parameters and can perform reliable sibilance detection and reduction regardless of absolute signal level, singer gender and other extraneous factors. An audio processing device divides input audio signals into buffers each containing a number of samples, the buffers overlapping one another. The audio processing device transforms each buffer from the time domain into the frequency domain and implements de-essing as a multi-band compressor that only acts on a designated sibilance band. The audio processing device determines an amount of attenuation in the sibilance band based on comparison of energy level in sibilance band of a buffer to broadband energy level in a previous buffer. The amount of attenuation is also determined based on a zero-crossing rate, as well as a slope and onset of a compression curve.

Disclosed is a method and apparatus for determining one or more operation parameters for a dynamic range compression (DRC) system. The method comprises obtaining, as an input, a parameter indicative of a hearing ability of a user, the parameter relating to a first difference in sound intensity between a maskee at a first frequency and a masker at a second frequency, determining a target value for the parameter, and determining the one or more operation parameters such that a second difference in sound intensity after sound intensity modification by the DRC (between sound intensity of the maskee of the masker) corresponds to the target value for the parameter. The operation parameters are determined such that a dependence of the second difference in sound intensity on the sound intensity of the maskee is minimized for a given range of sound intensities of the maskee.

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.

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.