Qualcomm Incorporated

S M Akramus Salehin

Siddhartha Goutham Swaminathan

Dipanjan Sen

Nils Günther Peters

Shankar Thagadur Shivappa

Isaac Garcia Munoz

Moo Young Kim

Erik Visser

Lae-Hoon Kim

Nils Gunther Peters

Shuhua Zhang

Sanghyun Chi

Ferdinando Olivieri

Jason Filos

Vasudev Nayak

Michael Ericson

Xiaoxin Zhang

Martin James Morrell

Ricardo De Jesus Bernal Castillo

Hannes Pessentheiner

Martin Morrell

Gang Chen

Methods, systems, computer-readable media, and apparatuses for manipulating a soundfield are presented. Some configurations include receiving a bitstream that comprises metadata and a soundfield description; parsing the metadata to obtain an effect identifier and at least one effect parameter value; and applying, to the soundfield description, an effect identified by the effect identifier. The applying may include using the at least one effect parameter value to apply the identified effect to the soundfield description.

Signalling of audio effect metadata in a bitstream

An apparatus includes one or more processors configured to receive orientation data and to select, based on the orientation data, a particular filter from among multiple filters. The one or more processors are configured to perform signal processing operations associated with three-dimensional (3D) sound data based on the particular filter.

3D sound orientation adaptability

A device includes a memory configured to store untransformed ambisonic coefficients at different time segments. The device includes one or more processors configured to obtain the untransformed ambisonic coefficients at the different time segments, where the untransformed ambisonic coefficients at the different time segments represent a soundfield at the different time segments. The one or more processors are configured to apply one adaptive network, based on a constraint that includes preservation of a spatial direction of one or more audio sources in the soundfield at the different time segments, to the untransformed ambisonic coefficients at the different time segments to generate transformed ambisonic coefficients at the different time segments, wherein the transformed ambisonic coefficients at the different time segments represent a modified soundfield at the different time segments, that was modified based on the constraint. The one or more processors are also configured to apply an additional adaptive network.

Transform ambisonic coefficients using an adaptive network for preserving spatial direction

An example audio decoding device includes a memory configured to store at least a portion of a coded audio bitstream; and one or more processors configured to: decode, based on the coded audio bitstream, a representation of a soundfield; decode, based on the coded audio bitstream, a syntax element indicating a selection of either a head-related transfer function (HRTF) or a binaural room impulse response (BRIR); and render, using the selected HRTF or BRIR, speaker feeds from the soundfield.

Signaling for rendering tools

A device and method for backward compatibility for virtual reality (VR), mixed reality (MR), augmented reality (AR), computer vision, and graphics systems. The device and method enable rendering audio data with more degrees of freedom on devices that support fewer degrees of freedom. The device includes memory configured to store audio data representative of a soundfield captured at a plurality of capture locations, metadata that enables the audio data to be rendered to support N degrees of freedom, and adaptation metadata that enables the audio data to be rendered to support M degrees of freedom. The device also includes one or more processors coupled to the memory, and configured to adapt, based on the adaptation metadata, the audio data to provide the M degrees of freedom, and generate speaker feeds based on the adapted audio data.

Six degrees of freedom and three degrees of freedom backward compatibility

A device may be configured to play one or more of a plurality of audio streams. The device may include a memory configured to store the plurality of audio streams, each of the audio streams representative of a soundfield. The device also may include one or more processors coupled to the memory, and configured to present a user interface to a user, obtain an indication from the user via the user interface representing a desired listening position, select, based on the indication, at least one audio stream of the plurality of audio streams, and output, for a display and in response to obtaining the indication representing the desired listening position, a graphical user interface element suggesting an alternative listening position.

User interface feedback for controlling audio rendering for extended reality experiences

A device includes one or more processors configured to determine, based on data descriptive of two or more audio environments, a geometry of a mutual audio environment. The one or more processors are also configured to process audio data, based on the geometry of the mutual audio environment, for output at an audio device disposed in a first audio environment of the two or more audio environments.

Augmented audio for communications

An example device includes a memory configured to store audio data and location data associated with a plurality of audio streams and one or more processors coupled to the memory. The one or more processors are configured to obtain a first location of a first audio stream that includes an audio source and obtain a second location of a second audio stream that includes the audio source. The one or more processors are configured to generate direction vectors originating at the first location and the second location, based on a location of the audio source and the first location, and the location of the audio source and the second location, respectively. The one or more processors are also configured to determine parameters that describe a vector field based on the first direction vector and the second direction vector.

Correlation-based rendering with multiple distributed streams accounting for an occlusion for six degree of freedom applications

A device includes one or more processors configured to obtain audio signals representing sound captured by at least three microphones and determine spatial audio data based on the audio signals. The one or more processors are further configured to determine a metric indicative of wind noise in the audio signals. The metric is based on a comparison of a first value and a second value. The first value corresponds to an aggregate signal based on the spatial audio data, and the second value corresponds to a differential signal based on the spatial audio data.

Spatial audio wind noise detection

A device includes a memory configured to store untransformed ambisonic coefficients at different time segments. The device also includes one or more processors configured to obtain the untransformed ambisonic coefficients at the different time segments, where the untransformed ambisonic coefficients at the different time segments represent a soundfield at the different time segments. The one or more processors are also configured to apply one adaptive network, based on a constraint, to the untransformed ambisonic coefficients at the different time segments to generate transformed ambisonic coefficients at the different time segments, wherein the transformed ambisonic coefficients at the different time segments represent a modified soundfield at the different time segments, that was modified based on the constraint.

Transform ambisonic coefficients using an adaptive network

An example device for processing one or more audio streams includes a memory configured to store the one or more audio streams and one or more processors implemented in circuitry coupled to the memory. The one or more processors are configured to determine a listener position. The one or more processors are also configured to determine one or more clusters of the one or more audio streams. The one or more processors are also configured to determine a rendering mode based on the listener position and the one or more clusters. The device also includes a renderer configured to render at least one of the one or more clusters of audio streams based on the rendering mode.

Smart hybrid rendering for augmented reality/virtual reality audio

In some examples, a content consumer device configured to play one or more of a plurality of audio streams includes a memory configured to store the plurality of audio streams and audio location information associated with the plurality of audio streams and representative of audio stream coordinates in an acoustical space where an audio stream was captured or synthesized or both. Each of the audio streams is representative of a soundfield. The content consumer device also includes one or more processors coupled to the memory, and configured to determine device location information representative of device coordinates of the content consumer device in the acoustical space. The one or more processors are configured to select, based on the device location information and the audio location information, a subset of the plurality of audio streams, and output, based on the subset of the plurality of audio streams, one or more speaker feeds.

Audio capture and rendering for extended reality experiences

A device may be configured to play one or more of a plurality of audio streams. The device may include a memory configured to store the plurality of audio streams, each of the audio streams representative of a soundfield. The device also may include one or more processors coupled to the memory, and configured to present a user interface to a user, obtain an indication from a user via the user interface representing a desired listening position; and select, based on the indication, at least one audio stream of the plurality of audio streams.

User interface for controlling audio rendering for extended reality experiences

In an aspect, a lens is zoomed in to create a zoomed lens. Lens data associated with the lens includes a direction of the lens relative to an object in a field-of-view of the zoomed lens and a magnification of the object resulting from the zoomed lens. An array of microphones capture audio signals including audio produced by the object and interference produced by other objects. The audio signals are processed to identify a directional component associated with the audio produced by the object and three orthogonal components associated with the interference produced by the other objects. Stereo beamforming is used to increase a magnitude of the directional component (relative to the interference) while retaining a binaural nature of the audio signals. The increase in magnitude of the directional component is based on an amount of the magnification provided by the zoomed lens to the object.

Spatial audio zoom

A device may be configured to process one or more audio streams in accordance with the techniques described herein. The device may comprise: one or more processors and a memory. The one or more processors may be configured to obtain an indication of a boundary separating an interior area from an exterior area, and obtain a listener location indicative of a location of the device relative to the interior area. The one or more processors may be configured to obtain, based on the boundary and the listener location, a current renderer as either an interior renderer configured to render audio data for the interior area or an exterior renderer configured to render the audio data for the exterior area, and apply, to the audio data, the current renderer to obtain one or more speaker feeds. The memory may be configured to store the one or more speaker feeds.

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San Diego, CA, United States of America

S M Akramus Salehin