Methods and systems for intuitive spatial audio rendering with improved intelligibility are disclosed. By establishing a virtual association between an audio source and a location in the listener's virtual audio space, a spatial audio rendering system can generate spatial audio signals that create a natural and immersive audio field for a listener. The system can receive the virtual location of the source as a parameter and map the source audio signal to a source-specific multi-channel audio signal. In addition, the spatial audio rendering system can be interactive and dynamically modify the rendering of the spatial audio in response to a user's active control or tracked movement.

One or more embodiments include an audio processing system for generating an audio scene for an extended reality (XR) environment. The audio processing system determines that a first virtual sound source associated with the XR environment affects a sound in the audio scene. The audio processing system generates a sound component associated with the first virtual sound source based on a contribution of the first virtual sound source to the audio scene. The audio processing system maps the sound component to a first loudspeaker included in a plurality of loudspeakers. The audio processing system outputs at least a first portion of the component for playback on the first loudspeaker.

The implementation of modal processors, which involve the parallel combination resonant filters, may be costly for applications such as artificial reverberation that can require thousands of modes. In one embodiment, the input signal is decomposed into a plurality of subbands, the outputs of which are downsampled. In each downsampled band, resonant filters are applied at the downsampled sampling rate, and their output is upsampled and filtered to form the band output. In these and other embodiments, a feature of responses of the mode filters have been optimized to minimize an aspect of a residual error after a point in time.

A sound field control apparatus includes a microphone configured to receive an utterance of a user, an output interface configured to output at least one of a sound signal and image data, and one or more processors configured to cancel a sound signal in a specific area around the microphone, obtain room impulse response information based on a user utterance position when the utterance of the user is received, and output a sound signal for providing an independent sound field to the user based on the room impulse response information.

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.

Some implementations involve receiving a content stream that includes audio data, receiving at least one type of level adjustment indication relating to playback of the audio data and controlling a level of the input audio data, based on the at least one type of level adjustment indication, to produce level-adjusted audio data. Some examples involve determining, based at least in part on the type(s) of level adjustment indication, a multiband limiter configuration, applying the multiband limiter to the level-adjusted audio data, to produce multiband limited audio data and providing the multiband limited audio data to one or more audio reproduction transducers of an audio environment.

A first endpoint generates an acoustic spread spectrum signal including a pilot sequence and a data sequence representing a token synchronized to the pilot sequence, transmits the acoustic spread spectrum signal, and records a transmit time at which the acoustic spread spectrum signal is transmitted. A receive time at which a second endpoint received the acoustic spread spectrum signal transmitted by the first endpoint is received from the second endpoint along with an indication of a second token as recovered from the received acoustic spread spectrum signal by the second endpoint. A separation distance between the first endpoint and the second endpoint is computed based on a time difference between the transmit time and the receive time. The first endpoint is paired with the second endpoint when the token matches the second token and the computed distance is less than a threshold distance.

An apparatus comprising means configured to: obtain at least one impulse response; obtain at least one reflection filter based on the obtained at least one impulse response, wherein the at least one reflection filter is configured to determine at least one early reflection from an acoustic surface which is not overlapped in time by any other reflection, wherein a duration of the at least one early reflection is shorter than a duration of the obtained at least one impulse response. In addition, an apparatus comprising means configured to: obtain at least one impulse response, wherein the at least one impulse response is configured with a perceivable timbre during rendering; create a timbral modification filter; obtain at least one audio signal; and render at least one output audio signal based n the at least one audio signal, wherein the at least one output signal is based on an application of the timbral modification filter.

A sound signal processing method includes: obtaining a sound signal; obtaining impulse response data that was measured in a predetermined space before the sound signal is obtained; generating an early reflected sound control signal not including a reverberant sound by convolving impulse response data of an early reflected sound among the obtained impulse response data into the obtained sound signal.

The invention relates to a method for improving the sound quality of a sound reproduction or recording in a room, the method comprising the steps of measuring an impulse response that comprises the linear response of the room; performing a time domain analysis to determine the resonances of the room and for a chosen group of room resonances determining a corresponding group of filters that, when inserted in a sound reproduction or recording chain in said room will counteract the unwanted effect of said chosen group of room resonances on the sound quality of sound reproduction or recording made in the room. The invention further relates to a device designed to implement the method according to the invention and to the use of a measure of amplitude decay as a function of frequency of a measured impulse response of a sound reproduction or recording system in a room to determine one or more resonance frequencies, the total or partial compensation of which will improve the sound quality of sound reproductions or recordings made in the room.