A circuit and method is disclosed for filtering an audio signal. The circuit has a first quadrature source and multipliers for multiplying the input signal by the I and Q outputs of the quadrature source. The multiplied inputs are then passed through a pair of low pass filters, which may have an adjustable Q factor. The outputs of the low pass filters are then multiplied in a second pair of multipliers by the I and Q outputs, respectively, of a second quadrature source, which will typically be of the same frequency, but different amplitude and phase, of the first quadrature source. The twice-multiplied signals are then summed by an adder to provide an output signal. The circuit may be modified to include a companding circuit between the low pass filters and the second pair of multipliers that determines the amplitude of the input signal, filters it, and compands the signal in a compandor. The compandor may have adjustable parameters. The circuit thus allows for far greater flexibility and control of the processing of the input signal than prior art circuits.

Methods, systems, and apparatuses are described herein for improved processing audio in a video stream. A system may split audio in a frame of video content into multiple bands based on their audio levels. The system may then dynamically compress and dynamically normalize the audio level in each band. When dynamically compressing the bands, the system may determine, based on stored information, what audio level range is acceptable for an end user and may smooth and maintain the ranges of the audio to be within the acceptable range. The system may include the dynamically normalized and dynamically compressed frames as a second audio track in the video content. A computing device receiving the video content may select the second audio track during playback. If an end user selects the second audio track, the video is delivered with the modified sound of the second audio track.

A cochlear implant is disclosed in an embodiment. The implant includes a signal level detector configured to determine a total signal level of an incoming acoustic signal and a processing unit configured to determine, in accordance with the determined total signal level, an intermediate knee point for each of a plurality of frequency bands. The implant includes a bandpass filters configured to generate a plurality of band limited audio signals in dependence upon the incoming acoustic signal, each band limited acoustic signal representing an associated audio frequency range relating to at least one electrode of a plurality of an implanted electrode array of the cochlear implant. The implant includes a pulse controller configured to deliver electrical stimulation signals to the plurality of electrodes of the implanted electrode array based on the generated signals and the determined intermediate knee point corresponding to the frequency range.

The present document describes a dynamic range control unit (210) configured to apply dynamic range control, referred to as DRC, to an audio signal (211). The DRC unit (210) is configured to downsample a subband signal (212) derived from the audio signal (211), to provide a downsampled subband signal (321), to determine a DRC gain (329) based on the downsampled subband signal (321), and to apply the DRC gain (329) to the subband signal (212), to provide a compressed subband signal (213) of a compressed audio signal (214).

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.

The invention provides a loudspeaker nonlinear compensation method. The method includes steps: obtaining system parameter of the loudspeaker, the No. i time-domain excitation voltage signal and the No. i state vector of the loudspeaker; compensating the No. i time-domain excitation voltage signal according to the system parameter and the No. i state vector and obtain i compensation voltage signal; obtaining the No. i+1 state vector according to the calculation of the system parameter and the No. i compensation voltage signal; outputting the No. i compensation voltage signal and record the quantity of the compensation voltage signal; judging whether the quantity of the compensation voltage signal is equal to the preset number value.

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.

An audio signal processing apparatus that includes a band division section, an analysis section, a gain adjustment amount calculation section, and a gain adjustment section. The band division section is configured to generate a resonance in-band signal by band division on an audio input signal. The analysis section is configured to extract an amount of features from each of the resonance in-band signal and the input signal. The gain adjustment amount calculation section is configured to calculate a gain adjustment amount for a resonance frequency band in the input signal, the gain adjustment amount being calculated based on the amount of features of the resonance in-band signal and the amount of features of the input signal. The gain adjustment section is configured to perform a gain adjustment on the resonance frequency band in the input signal based on the gain adjustment amount.

A hearing aid includes: an input transducer for provision of an audio signal in response to sound; a hearing loss model for calculation of a hearing loss as a function of a signal level of the audio signal; and a probabilistic hearing loss compensator that is configured to process the audio signal into a hearing loss compensated audio signal in such a way that the hearing loss is restored to normal hearing in accordance with the hearing loss model.

An example method includes receiving data indicating a configuration of one or more playback devices. The one or more playback devices may include one or more transducers. The method further includes, based on the received data, associating each of one or more audio streams respectively with at least one transducer of the one or more transducers. The method further includes generating the one or more audio streams and sending at least one of the generated one or more audio streams to each of the one or more playback devices. An example non-transitory computer readable medium and an example computing device related to the example method are also disclosed herein.