A computer-implemented method of audio processing, the method comprising: receiving audio object data and audio description data, wherein the audio object data includes a first plurality of audio objects; calculating a long-term loudness of the audio object data and a long- term loudness of the audio description data; calculating a plurality of short-term loudnesses of the audio object data and a plurality of short-term loudnesses of the audio description data; reading a first plurality of mixing parameters that correspond to the audio object data; generating a second plurality of mixing parameters based on the first plurality of mixing parameters, the long-term loudness of the audio object data, the long-term loudness of the audio description data, the plurality of short-term loudnesses of the audio object data, and the plurality of short-term loudnesses of the audio description data; generating a gain adjustment visualization corresponding to the second plurality of mixing parameters, the audio object data and the audio description data; and generating mixed audio object data by mixing the audio object data and the audio description data according to the second plurality of mixing parameters, wherein the mixed audio object data includes a second plurality of audio objects, wherein the second plurality of audio objects correspond to the first plurality of audio objects mixed with the audio description data according to the second plurality of mixing parameters.

Multiple virtual source locations may be defined for a volume within which audio objects can move. A set-up process for rendering audio data may involve receiving reproduction speaker location data and pre-computing gain values for each of the virtual sources according to the reproduction speaker location data and each virtual source location. The gain values may be stored and used during “run time,” during which audio reproduction data are rendered for the speakers of the reproduction environment. During run time, for each audio object, contributions from virtual source locations within an area or volume defined by the audio object position data and the audio object size data may be computed. A set of gain values for each output channel of the reproduction environment may be computed based, at least in part, on the computed contributions. Each output channel may correspond to at least one reproduction speaker of the reproduction environment.

A system and a method for outputting sound where one or more first sound outputting units are identified and other, second, sound outputting units define a sound delay in accordance with the relative positions between each sound outputting unit and the first sound outputting unit(s). Microphones may be added to e.g. determine the amount and positions of persons.

The system may also use Intelligent cameras to determine number of people their position the room face direction and age distribution in order optimize audio level and equalisation of the frequency response—like in a church with many elder people or in a young audience at a live concert.

An apparatus configured to: obtain at least one spatial audio signal that defines an audio scene forming at least in part an immersive media content; obtain metadata associated with the at least one spatial audio signal; obtain at least one augmentation control parameter associated with the at least one spatial audio signal; obtain at least one augmentation audio signal; render an output audio signal that is based, at least partially, on the at least one spatial audio signal, the metadata associated with the at least one spatial audio signal, the at least one augmentation control parameter, and the at least one augmentation audio signal; and obtain an indication that at least part of the at least one spatial audio signal has been omitted from the output audio signal based, at least partially, on at least part of the at least one augmentation audio signal included in the output audio signal.

An audio system presented herein includes a transducer array, a sensor array, and a controller. The transducer array presents audio content to a user. The controller controls the transducer array to adjust a level of tactile content imparted to the user via actuation of at least one transducer in the transducer array while presenting the audio content to the user. The audio system can be part of a headset.

There is provided encoding and decoding methods for encoding and decoding of object based audio. An exemplary encoding method includes inter alia calculating M downmix signals by forming combinations of N audio objects, wherein M≤N, and calculating parameters which allow reconstruction of a set of audio objects formed on basis of the N audio objects from the M downmix signals. The calculation of the M downmix signals is made according to a criterion which is independent of any loudspeaker configuration.

Disclosed are methods and systems which convert a multi-microphone input signal to a multichannel output signal making use of a time- and frequency-varying matrix. For each time and frequency tile, the matrix is derived as a function of a dominant direction of arrival and a steering strength parameter. Likewise, the dominant direction and steering strength parameter are derived from characteristics of the multi-microphone signals, where those characteristics include values representative of the inter-channel amplitude and group-delay differences.

A method includes generating a synthesized non-reference high-band channel based on a non-reference high-band excitation corresponding to a non-reference target channel. The method further includes estimating one or more spectral mapping parameters based on the synthesized non-reference high-band channel and a high-band portion of the non-reference target channel. The method also includes applying the one or more spectral mapping parameters to the synthesized non-reference high-band channel to generate a spectrally shaped synthesized non-reference high-band channel. The method further includes generating an encoded bitstream based on the one or more spectral mapping parameters and the spectrally shaped synthesized non-reference high-band channel.

An audio system includes an equatorial acoustic sensor array (EASA) that may be coupled to an object. The audio system is configured to detect, via the EASA, signals corresponding to a portion of a sound field in a local area. The detected signals are converted into a plurality of corresponding abstract representations that describe the portion of the sound field. Effects of scattering of the object are removed from the abstract representations to create adjusted abstract representations. A set of spherical harmonic (SH) coefficients is determined using the adjusted abstract representations. The set of SH coefficients describe an entirety of the sound field. And the set of SH coefficients and head related transfer functions of a user are used for binaural rendering of the reconstructed sound field to the user.

A sound spatialisation method includes determining digital processing parameters to be applied to sound signals to be broadcast by a set of at least two loudspeakers in order to reproduce a virtual sound source at a desired position, and restoring sound signals by the loudspeakers during which the digital processing parameters are applied to the sound signals. The sound spatialisation method also includes defining a trajectory defined by a set of N points, with two consecutive points of the trajectory being connected together by a curve, and positioning during which the desired position of the virtual sound source is defined on the trajectory.