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.

An apparatus for processing soundfield data is provided. The soundfield data defines a soundfield within a spatial reproduction region comprising at least one bright zone and at least one quiet zone. The apparatus comprises an applicator configured to apply a spatially continuously varying weighting function to the soundfield data in order to obtain weighted soundfield data defining a weighted soundfield, wherein the spatially continuously varying weighting function is configured to enhance the soundfield in at least one of the bright zone and the quiet zone.

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.

A sound field device is disclosed that comprises an elevation cue estimator, a low-frequency filter estimator, and a high-frequency filter estimator. The elevation cue-estimator is configured to estimate an elevation cue of a head-related transfer function (HRTF) of at least one listener. The low-frequency filter estimator is configured to estimate one or more low-frequency filter elements based on the elevation cue. The high-frequency filter estimator is configured to estimate one or more high-frequency filter elements based on the elevation cue. An estimation method of the low-frequency filter estimator is different from an estimation method of the high-frequency filter estimator. The one or more low-frequency filter elements and the one or more high-frequency filter elements are for driving an array of loudspeakers to generate an elevated sound impression at a bright zone.

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.

The invention refers to a system for handling digital content including an input interface, a calculator, and an output interface. The input interface receives digital content and includes a plurality of input channels. At least one input channel receives digital content from a sensor or a group of sensors belonging to a recording session. The calculator provides output digital content by adapting received digital content to a reproduction session in which the output digital content is to be reproduced. The output interface outputs the output digital content and includes a plurality of output channels, wherein at least one output channel outputs the output digital content to an actuator or a group of actuators belonging to the reproduction session. Further, the input interface, the calculator, and the output interface are connected with each other via a network. The input interface is configured to receive digital content via by Ni input channels, where the number Ni is based on a user interaction, and/or the output interface is configured to output the output digital content via by No output channels, where the number No is based on a user interaction. The invention further refers to a corresponding method.

A sound processing apparatus includes a signal processing device that individually performs first localization setting (e.g., sound volume panning) for setting localization of an input sound signal based on a value of a first parameter and second localization setting (e.g., delay panning) for setting localization of the input sound signal based on a value of a second parameter. In response to an adjustment by an operation device of the value of one of the first and second parameters, a control device automatically changes the value of the other of the first and second parameters. In this way, sound image localization based on the first localization setting and sound image localization based on the second localization setting are automatically controlled in an interlocked relation to each other. At least one of the sound signals localized based on the first and/or second localization setting is output to an output destination.

A wave field synthesis apparatus for driving an array of loudspeakers with drive signals, the apparatus includes a sound field synthesizer for generating sound field drive signals for causing the array of loudspeakers to generate one or more sound fields at one or more audio zones, a binaural renderer for generating binaural drive signals for causing the array of loud-speakers to generate specified sound pressures at at least two positions, wherein the at least two positions are determined based on a detected position and/or orientation of a listener, and a decision unit for deciding whether to generate the drive signals using the sound field synthesizer or using the binaural renderer.

The present technology relates to a sound field forming apparatus and method and a program that are configured to enhance the reproducibility of a wavefront at a listener position. The sound field forming apparatus has a position acquisition unit configured to acquire position information indicative of a position of a listener or a position of a sound source to be formed, a control point specification unit configured to specify a control point in accordance with a distance from a speaker array of the listener or the sound source on the basis of the position information, and a filter unit configured to generate a speaker drive signal for forming a predetermined sound field by the speaker array by convoluting a filter coefficient corresponding to the specified control point with a sound source signal. The present technology can be applied to the sound field forming apparatus.

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.