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.

A device for calculating loudspeaker signals for a plurality of loudspeakers while using a plurality of audio sources, an audio source including an audio signal, includes a forward transform stage for transforming each audio signal, block-by-block, to a spectral domain so as to obtain for each audio signal a plurality of temporally consecutive short-term spectra, a memory for storing a plurality of temporally consecutive short-term spectra for each audio signal, a memory access controller for accessing a specific short-term spectrum among the plurality of short-term spectra for a combination consisting of a loudspeaker and an audio signal on the basis of a delay value, a filter stage for filtering the specific short-term spectrum for the combination of the audio signal and the loudspeaker by using a filter provided for the combination of the audio signal and the loudspeaker, so that a filtered shot-term spectrum is obtained for each combination of an audio signal and a loudspeaker, a summing stage for summing up the filtered short-term spectra for a loudspeaker so as to obtain summed-up short-term spectra for each loudspeaker, and a backtransform stage for backtransforming, block-by-block, summed-up short-term spectra for the loudspeakers to a time domain so as to obtain the loudspeaker signals.

An information processing device includes circuitry that detects a change in a position of a person. The circuitry also changes a position of a virtual sound source in accordance with the change in the position of the person and outputs a sound through wave field synthesis so that a spatial sound field is transmitted via a plurality of speakers.

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 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 best match 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 based on a result of the RIR comparison.

A signal processing apparatus includes: an audio signal processing unit configured to perform wavefront synthesis processing for at least part of a plurality of sound source data; a first output unit configured to output N-channel audio signals output from the audio signal processing unit to a first speaker device; a mix processing unit configured to mix the N-channel audio signals output from the audio signal processing unit; and a second output unit configured to output an audio signal output from the mix processing unit to a second speaker device, in which a setting regarding an output of the second speaker device is possible.

An audio control system includes at least one microphone, at least one speaker, and one or more processors. The at least one microphone is at a first position adjacent to an ear of a user and configured to detect a sound field at the first position and output a sound signal indicative of the detected sound field. The at least one speaker is at a second position spaced from the first position and configured to output sound responsive to receiving an audio signal. The one or more processors are configured to generate the audio signal based on the sound signal and a target parameter of the sound field at (i) the first position and (ii) a third position spaced from the first position; and provide the audio signal to the at least one speaker to cause the at least one speaker to output the sound responsive to receiving the audio signal.

An audio processing apparatus comprises a receiver (705) which receives audio data including audio components and render configuration data including audio transducer position data for a set of audio transducers (703). A renderer (707) generating audio transducer signals for the set of audio transducers from the audio data. The renderer (7010) is capable of rendering audio components in accordance with a plurality of rendering modes. A render controller (709) selects the rendering modes for the renderer (707) from the plurality of rendering modes based on the audio transducer position data. The renderer (707) can employ different rendering modes for different subsets of the set of audio transducers the render controller (709) can independently select rendering modes for each of the different subsets of the set of audio transducers (703). The render controller (709) can select the rendering mode for a first audio transducer of the set of audio transducers (703) in response to a position of the first audio transducer relative to a predetermined position for the audio transducer. The approach may provide improved adaptation, e.g. to scenarios where most speakers are at desired positions whereas a subset deviate from the desired position(s).

The present technology relates to a signal processing device, signal processing method, and program capable of providing a higher realistic feeling.

A signal processing device includes: an acquisition unit that acquires audio data of an audio object and metadata including position information indicating a position of the audio object and direction information indicating a direction of the audio object; and a signal generation unit that generates a reproduction signal for reproducing a sound of the audio object at a listening position on the basis of listening position information indicating the listening position, listener direction information indicating a direction of a listener at the listening position, the position information, the direction information, and the audio data. The present technology is applicable to a transmission reproduction system.

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 following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals, which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.