An audio system includes at least one sound generation device having at least an acoustic transducer; and an electronic integrated circuit. A communication bus is connected to the at least one sound generation device and is configured to communicate a digital signal comprised of one or more audio streams and control signals. A power bus is connected to the at least one sound generation device. The acoustic transducer includes at least one of the following groups: a membrane and an acoustic modulator; or a membrane, acoustic resonator, and acoustic coupler. The electronic integrated circuit is constructed and arranged to receive the digital signal and generate an analog electric signal to operate the acoustic transducer to generate an audio signal in accordance with the control signal.

In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

An apparatus and method of rendering audio. A binaural signal is split on an amplitude weighting basis into a front binaural signal and a rear binaural signal, based on perceived position information of the audio. In this manner, the front-back differentiation of the binaural signal is improved.

An apparatus, method and computer program is described including: determining one of a plurality of audio modes for presentation of first audio to a user based on a location and/or movement of the user within a virtual scene, wherein the first audio includes a plurality of audio tracks located at different locations within the virtual scene; and rendering the first audio to the user in the determined audio mode, wherein: in a first audio mode of the plurality of audio modes, the locations of the audio tracks within the virtual scene are fixed and in a second audio mode of the plurality of audio modes, the locations of the audio tracks within the virtual scene move with the user.

A method of playing back audio content with improved immersiveness can include receiving, at a playback device, audio input including vertical content having a high-frequency portion and a low-frequency portion. The playback device can face along a first sound axis and comprise an up-firing transducer configured to direct sound along a second sound axis that is vertically angled with respect to the primary sound axis and a side-firing transducer or array configured to direct sound along a third axis that is horizontally angled with respect to the first sound axis. The low-frequency portion of the vertical content can be played back via the side-firing transducer or array, while the high-frequency portion of the vertical content can be played back via the up-firing transducer.

An audio processing system and method which calculates, based on spatial metadata of the audio object, a panning coefficient for each of the audio objects in relation to each of a plurality of predefined channel coverage zones. Converts the audio signal into submixes in relation to the predefined channel coverage zones based on the calculated panning coefficients and the audio objects. Each of the submixes indicating a sum of components of the plurality of the audio objects in relation to one of the predefined channel coverage zones. Generating a submix gain by applying an audio processing to each of the submix and controls an object gain applied to each of the audio objects. The object gain being as a function of the panning coefficients for each of the audio objects and the submix gains in relation to each of the predefined channel coverage zones.

Simulating a 3D audio environment, including receiving a visual representation of an object at a location in a scene, wherein the location represents a point in 3D space, receiving a sound element, and binding the sound element to the visual representation of the object such that a characteristic of the sound element is dynamically modified coincident with a change in location in the scene of the visual representation of the object in 3D space.

An apparatus, method and computer program product for receiving captured spatial audio information comprising at least one audio signal and at least one audio parameter for controlling the at least one audio signal, receiving an instruction to set an audio parameter to a first parameter value, setting, in response to determining that the first parameter value fulfils at least one criterion, the audio parameter to a second parameter value and changing the audio parameter from the second parameter value to the first parameter value.

A sound signal downmix method includes an inter-channel relationship information obtaining step of obtaining an inter-channel correlation value and an inter-channel time difference in an approximate manner, and a downmix step of obtaining a downmix signal based on the obtained information. In the inter-channel relationship information obtaining step, multiple channel signals are sorted such that signals of adjacent channels are similar to each other, the inter-channel correlation value and the inter-channel time difference are determined only between adjacent channels after the sorting, the inter-channel correlation value between non-adjacent channels is obtained by a product or a geometric mean of the inter-channel correlations between the adjacent channels, and the inter-channel time difference between non-adjacent channels is obtained by adding up the inter-channel time differences of adjacent channels.

An apparatus including circuitry configured to: obtain two or more audio signals from respective two or more microphones; determine, in one or more frequency band of the two or more audio signals, a first sound source direction parameter based on processing of the two or more audio signals, wherein processing of the two or more audio signals is further configured to provide one or more modified audio signal based on the two or more audio signals; and determine, in the one or more frequency band of the two or more audio signals, at least a second sound source direction parameter at least based on at least in part the one or more modified audio signal.