A method for processing an audio signal comprising a first and second spatialized audio signal, wherein the first audio signal has a first lateral spatializer of a multichannel audio spatializer, the second spatialized audio signal (RI) a second lateral spatializer of the multichannel audio spatializer, and the first and second spatialized audio signals differs from the each other. Also disclosed is a first equalizer transfer function receiving and filtering the first spatialized audio signal from first set of equalizer coefficients to provide a first equalized audio signal and second equalizer transfer function receiving and filtering the second spatialized audio signal.

An electronic furniture assembly of the present invention includes: (i) a furniture assembly comprising: (A) a base (e.g., a seat portion), (B) at least one transverse or upright member (e.g., a side, armrest or backrest), and (ii) a speaker system mounted within one or more portions of the furniture assembly. A method of tuning a speaker of an audio-enhanced furniture system includes presenting a user with predetermined tuning profiles corresponding to a plurality of configuration footprint shapes in which the furniture assembly can be assembled and in response to a user selection of a given configuration footprint shape, tuning the speaker system according to the tuning profile selected by the user.

A speaker device includes left and right speakers and a correction part. The correction part corrects frequency characteristics of a stereo audio signal supplied to the left and right speakers so that transfer characteristics of the left and right speakers become predetermined transfer characteristics.

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.

Methods, systems, and computer program products for rending an audio object having an apparent size are disclosed. An audio processing system receives audio panning data including a first grid mapping first virtual sound sources in a space and speaker positions to speaker gains. The first grid specifies first speaker gains of the first virtual sound sources in the space. The audio processing system determines a second grid of second virtual sound sources in the space, including mapping the first virtual sound sources into the second virtual sound sources of the second virtual sources. The audio processing system selects at least one of the first grid or second grid for rendering an audio object based on an apparent size of the audio object. The audio processing system renders the audio object based on the selected grid or grids.

A device can include a microphone, a memory configured to store a sound source table including correspondence relationships among boarding items, virtual sound items and filter set values, and a processor configured to receive a sound signal through the microphone and determine a first filter set value matching characteristics of the received sound signal using the sound source table.

Systems and methods are disclosed for providing voice and noise activity detection with audio automixers that can reject errant non-voice or non-human noises while maximizing signal-to-noise ratio and minimizing audio latency.

In some embodiments, virtualization methods for generating a binaural signal in response to channels of a multi-channel audio signal, which apply a binaural room impulse response (BRIR) to each channel including by using at least one feedback delay network (FDN) to apply a common late reverberation to a downmix of the channels. In some embodiments, input signal channels are processed in a first processing path to apply to each channel a direct response and early reflection portion of a single-channel BRIR for the channel, and the downmix of the channels is processed in a second processing path including at least one FDN which applies the common late reverberation. Typically, the common late reverberation emulates collective macro attributes of late reverberation portions of at least some of the single-channel BRIRs. Other aspects are headphone virtualizers configured to perform any embodiment of the method.

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.

Embodiments of this application provide an audio data processing method and apparatus, and a sound box system, which are related to the field of audio technologies, and improve sound quality of an audio playing device. The sound box system includes a full-frequency sound box and a low-frequency sound box. The full-frequency sound box is physically connected to the low-frequency sound box by using a first fastening part of the full-frequency sound box and a second fastening part of the low-frequency sound box. Furthermore, the full-frequency sound box communicates with the low-frequency sound box by using a first communication part of the first fastening part and a second communication part of the second fastening part, where the first fastening part and the second fastening part are a group of paired connection parts, and the first communication part and the second communication part are a group of paired communication parts.