Embodiments are described for an adaptive audio system that processes audio data comprising a number of independent monophonic audio streams. One or more of the streams has associated with it metadata that specifies whether the stream is a channel-based or object-based stream. Channel-based streams have rendering information encoded by means of channel name; and the object-based streams have location information encoded through location expressions encoded in the associated metadata. A codec packages the independent audio streams into a single serial bitstream that contains all of the audio data. This configuration allows for the sound to be rendered according to an allocentric frame of reference, in which the rendering location of a sound is based on the characteristics of the playback environment (e.g., room size, shape, etc.) to correspond to the mixer's intent. The object position metadata contains the appropriate allocentric frame of reference information required to play the sound correctly using the available speaker positions in a room that is set up to play the adaptive audio content.

A lighting device comprising: a plurality of light emitting devices arranged in a two-dimensional array; a plurality of audio emitting devices co-located with the light emitting devices; and an optically translucent surface located forward of both the light emitting devices and the audio emitting devices such that the devices are not directly visible through the surface, wherein the surface is acoustically transparent such that sounds emitted from the audio emitting devices are audible through the surface; wherein the light emitting devices are controllable to render light effects at different locations on the surface, and the audio emitting devices are controllable to emit sounds perceived to originate from matching locations.

A method including receiving an audio scene including at least one source captured using at least one near field microphone and at least one far field microphone. The method includes determining at least one room-impulse-response (RIR) associated with the audio scene based on the at least one near field microphone and the at least one far field microphone, accessing a predetermined scene geometry corresponding to the audio scene, and identifying a best matching geometry to the predetermined scene geometry in a scene geometry database. The method also includes performing RIR comparison based on the at least one RIR and at least one geometric RIR associated with the best matching geometry, and rendering a volumetric audio scene experience based on a result of the RIR comparison.

The present disclosure relates to a signal processing device, a signal processing method, and a program that allows for generating an input signal suitable for a multi-way speaker. A band dividing unit divides an audio signal into signals in a plurality of bands corresponding to respective bands of a plurality of speaker units of the multi-way speaker. A filter processing unit performs wave front synthesis filter processing on each of the audio signals in the respective bands having been divided into. The audio signal in each of the bands after the wave front synthesis filter processing is supplied to the speaker unit of the corresponding band in the multi-way speaker. The present disclosure is applicable, for example, to a signal processing device or other devices.

A local wave field synthesis apparatus, which includes a determination module for determining desired sound pressures and desired particle velocity vectors at a plurality of control points, a computation module for computing sound pressures and particle velocity vectors at the plurality of control points based on a set of filter parameters, an optimization module for computing an optimum set of filter parameters by jointly optimizing computed sound pressures towards the desired sound pressures and computed particle velocity vectors towards the desired particle velocity vectors, and a generator module for generating the drive signals based on the optimum set of filter parameters, wherein the plurality of control points are located on one or more contours around the one or more audio zones.

Provided is a signal processing device including a display control unit for causing a display to display an image corresponding to a specified place, a sound-collection-signal input unit for inputting a sound collection signal of a sound collection unit that collects a user sound produced with microphones surrounding the user, an acoustic-signal processing unit for performing a first acoustic-signal process for reproducing a sound field where the user sound is sensed as if the sound were echoing in the place on the signal input by the sound-collection-signal input unit, based on a first transfer function measured in the place to indicate how a sound emitted on a closed surface inside the place echoes in the place and then is transferred to the closed-surface side, and a sound-emission control unit for causing a sound based on the processed signal to be emitted from speakers surrounding the user.

Higher Order Ambisonics represents three-dimensional sound independent of a specific loudspeaker set-up. However, transmission of an HOA representation results in a very high bit rate. Therefore compression with a fixed number of channels is used, in which directional and ambient signal components are processed differently. The ambient HOA component is represented by a minimum number of HOA coefficient sequences. The remaining channels contain either directional signals or additional coefficient sequences of the ambient HOA component, depending on what will result in optimum perceptual quality. This processing can change on a frame-by-frame basis.

Embodiments are described for an adaptive audio system that processes audio data comprising a number of independent monophonic audio streams. One or more of the streams has associated with it metadata that specifies whether the stream is a channel-based or object-based stream. Channel-based streams have rendering information encoded by means of channel name; and the object-based streams have location information encoded through location expressions encoded in the associated metadata. A codec packages the independent audio streams into a single serial bitstream that contains all of the audio data. This configuration allows for the sound to be rendered according to an allocentric frame of reference, in which the rendering location of a sound is based on the characteristics of the playback environment (e.g., room size, shape, etc.) to correspond to the mixer's intent. The object position metadata contains the appropriate allocentric frame of reference information required to play the sound correctly using the available speaker positions in a room that is set up to play the adaptive audio content.

The voice control device includes a sound source signal input unit, a frequency determination unit, a band controller, a sound image controller, and a voice output unit. The sound source signal input unit inputs a sound source signal of content from a sound source. The frequency determination unit determines a cutoff frequency. The band controller acquires a high frequency signal in a frequency band equal to or higher than the cutoff frequency and a low frequency signal in a frequency band equal to or lower than the cutoff frequency, from the sound source signal of the content. The sound image controller generates a plurality of sound image control signals for controlling sound images of the plurality of speakers, by controlling at least one of a phase and a sound pressure level of the high frequency signal. The voice output unit outputs the low frequency signal to a first speaker, and outputs the plurality of sound image control signals to a second speaker composed of a plurality of speakers.

Provided is a signal processing device including: a reverberation sound signal generation unit that generates a reverberation sound signal according to a sound source position of a virtual sound source and a distance to a reference point; and a drive signal generation unit that generates a drive signal for a speaker array by a wavefront synthesis filter, in which the drive signal generation unit generates the drive signal on the basis of a signal obtained by performing wavefront synthesis filtering processing on a signal obtained by convolving the reverberation sound signal with a signal of the virtual sound source and/or a signal obtained by performing wavefront synthesis filtering processing on the reverberation sound signal to make the reverberation sound signal into a virtual sound source.