There is provided a signal processing apparatus advantageous in terms of sound source separation performance. The signal processing apparatus includes a dividing unit configured to divide audio signal acquired by a plurality of audio acquisition units into components of a plurality of different frequency bands, and a processing unit configured to form, based on the audio signal, a plurality of directional beams having different directivities in accordance with a target direction and a target width. Each of the plurality of directional beams has directivities in different directions for the respective components of the frequency bands divided by the dividing unit.

There is provided encoding and decoding methods for encoding and decoding of object based audio. An exemplary encoding method includes inter alia calculating M downmix signals by forming combinations of N audio objects, wherein M≦N, and calculating parameters which allow reconstruction of a set of audio objects formed on basis of the N audio objects from the M downmix signals. The calculation of the M downmix signals is made according to a criterion which is independent of any loudspeaker configuration.

The present invention relates to a method for generating a filter for an audio signal and a parameterization device for the same, and more particularly, to a method for generating a filter for an audio signal, to implement filtering of an input audio signal with a low computational complexity, and a parameterization device therefor. To this end, provided are a method for generating a filter for an audio signal, including: receiving at least one binaural room impulse response (BRIR) filter coefficients for binaural filtering of an input audio signal; converting the BRIR filter coefficients into a plurality of subband filter coefficients; obtaining average reverberation time information of a corresponding subband by using reverberation time information extracted from the subband filter coefficients; obtaining at least one coefficient for curve fitting of the obtained average reverberation time information; obtaining flag information indicating whether the length of the BRIR filter coefficients in a time domain is more than a predetermined value; obtaining filter order information for determining a truncation length of the subband filter coefficients, the filter order information being obtained by using the average reverberation time information or the at least one coefficient according to the obtained flag information and the filter order information of at least one subband being different from filter order information of another subband; and truncating the subband filter coefficient by using the obtained filter order information and a parameterization device therefor.

Embodiments relate to audio processing for opposite facing speaker configurations that results in multiple optimal listening regions around the speakers. A system includes a left speaker and a right speaker in an opposite facing speaker configuration, and a crosstalk cancellation processor connected with the left speaker and the right speaker. The crosstalk cancellation processor applies a crosstalk cancellation to an input audio signal to generate left and right output channels. The left output channel is provided to the left speaker and the right output channel is provided to the right speaker to generate sound including multiple crosstalk cancelled listening regions that are spaced apart.

An apparatus for controlling a dynamic compressor of a hearing aid includes a combination signal analyzer for determining the binaural similarity of a left and right audio signal and an amplification adjuster for providing an amplification value for a band of the left or right audio signal in dependence on the binaural similarity and a level of the left or right audio signal in the band.

An acoustic device includes: a generating unit that generates a monaural signal on the basis of a left stereo signal and a right stereo signal in a low frequency band; an extracting unit that extracts a stereo component for an L-channel and a stereo component for an R-channel on the basis of a left stereo signal and a right stereo signal in a high frequency band; a first combining unit that combines the monaural signal and the stereo component for the L-channel; and a second combining unit that combines the monaural signal and the stereo component for the R-channel.

In at least one embodiment, an audio system is provided. The audio system includes a plurality of loudspeaker, a plurality of microphones, and an audio controller. The plurality of loudspeakers transmits an audio signal in a listening environment. The plurality of microphones detects the audio signal in the listening environment. The at least one audio controller is configured to determine a first psychoacoustic perceived loudness (PPL) of the audio signal as the audio signal is played back through a first loudspeaker of the plurality of loudspeakers and to determine a second PPL of the audio signal as the audio signal is sensed by a first microphone of the plurality of microphones. The at least one audio controller is further configured to map the first loudspeaker of the plurality of loudspeakers to the first microphone of the plurality of microphones based at least on the first PPL and the second PPL.

A hearing system includes one or more hearing devices configured to be worn by a user. Each hearing device includes a signal source that provides an input electrical signal representing a sound of a virtual source. A filter implements a head related transfer function (HRTF) to add spatialization cues associated with a virtual location of the virtual source to the electrical signal and outputs a filtered electrical signal that includes the spatialization cues. A speaker of the hearing device converts the filtered electrical signal into an acoustic signal and plays the acoustic signal to the user. The system includes motion tracking circuitry that tracks motion of the user as the user moves in a direction of a perceived location that the user perceives to be the virtual location of the virtual source. Head related transfer function (HRTF) individualization circuitry determines a difference between the virtual location and the perceived location in response to the motion of the user. The HRTF individualization circuitry individualizes the HRTF based on the difference.

M audio signals are obtained by processing an audio signal by M virtual speakers; M first HRTFs and M second HRTFs are obtained, where the M first HRTFs corresponding to a left ear position, and the M second HRTFs corresponding to a right ear position; high-band impulse responses of some of the M first HRTFs are modified to obtain modified first target HRTFs, and high-band impulse responses of some of the M second HRTFs are modified to obtain modified second target HRTFs; a first target audio signal corresponding to the left ear position is obtained based on the modified first target HRTFs and un-modified first HRTFs, and the M audio signals; and a second target audio signal corresponding to the right ear position is obtained based on the modified second HRTFs, un-modified second target HRTFs, and the M audio signals.

An audio system is proposed, dynamically playing optimized audio signals based on user position. A sensor circuits dynamically senses a target space to generate field context information. First speaker and second speaker are arranged for audio playback. A host device recognizes a user from the field context information, determines the user position corresponding to the target space, and adaptively assigns the user position as a target listening spot. A sensor circuit contains a camera capturing an ambient image out of the target space. A recognizer circuit analyzes the ambient image to obtain from the target space, the location, size and acoustic attribute information of an ambient object, allowing the control circuit to accordingly perform an object-based compensation operation on the target listening spot to generate optimized first channel audio signal and second channel audio signal.