A method for audio rendering by an apparatus comprising at least one audio rendering device, the method comprising: extracting (S10) a plurality of frequency band components from an input audio signal; determining (S15) from the device frequency response and the plurality of extracted frequency band components at least one indicator representative of masking frequencies energy, masking frequencies corresponding to frequency bands that are above a frequency threshold; determining at least one correction factor (S16) from the indicator representative of masking frequencies energy; e) for each frequency band, determining a second acoustic level threshold (S20) by modifying (S17) with the correction factor a predetermined first acoustic level threshold associated with said frequency band, and determining a reduction gain from a comparison (S30) between an acoustic level of the extracted frequency band and the second acoustic level threshold associated to said extracted frequency band, and applying (S40) the reduction gain.

Equipment that can comfortably hear mixed large sound and small sound at the same time by earphones and headphones is achieved. Level of rise and fall of an input signal waveform is logarithmically compressed and output to limit wave height, and an output signal is taken out from a wave-height-limited signal by eliminating or attenuating harmonic distortion. A limiter 6 limits an amplitude of an input sound signal. The integrator 7 is connected to the output-side of the limiter 6 and logarithmically compresses rise or fall of an output signal waveform from the limiter 6.

Equipment that can comfortably hear mixed large sound and small sound at the same time by earphones and headphones is achieved. Level of rise and fall of an input signal waveform is logarithmically compressed and output to limit wave height, and an output signal is taken out from a wave-height-limited signal by eliminating or attenuating harmonic distortion. A limiter 6 limits an amplitude of an input sound signal. The integrator 7 is connected to the output-side of the limiter 6 and logarithmically compresses rise or fall of an output signal waveform from the limiter 6.

An audio system presented herein includes a transducer array, a sensor array, and a controller. The transducer array presents audio content to a user. The controller controls the transducer array to adjust a level of tactile content imparted to the user via actuation of at least one transducer in the transducer array while presenting the audio content to the user. The audio system can be part of a headset.

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.

In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

An information processing device (100) including a control unit that causes, on a basis of vibration output characteristics of a vibration device (200), the vibration device (200) to output a vibration corresponding to sound data.

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.

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.