Dolby Laboratories Licensing Corporation

Dolby International Ab

Lake Technology Limited

Lake Dsp Pty Limited

Dobly Laboratories Licensing Corporation

David S McGrath

Glenn N Dickins

Richard James Cartwright

Nicolas R Tsingos

Antonio Mateos Sole

Juan Felix Torres

Freddie Sanchez

Stefan Bruhn

Rhonda J Wilson

Michael Eckert

Andrew Peter Reilly

Adam Richard McKeag

Heiko Purnhagen

Dirk Jeroen Breebaart

Kristofer Kjoerling

Michael John Smithers

David Matthew Cooper

Michael William Mason

Stephen James Bennett

Stefanie Brown

Mark Franklin Davis

Harald Mundt

Regunathan Radhakrishnan

Zhiwei Shuang

Glen Norman Dickins

Leonard Layton

Nigel Lloyd William Heyler

Geoffrey Alexander Drane

Ming Wang

Brett Graham Crockett

Mark Stuart Vinton

Louis Dunn Fielder

Christophe Chabanne

Vinay Melkote

Mark R P Thomas

Gary Spittle

David D Clementson

Paul Francis Holmberg

David Matthew Fischer

Timothy James Eggerding

Adam Christopher Noel

Brian George Arnott

Hadis Nosrati

Eugene Edward Radzik

Daniel Heckenberg

Brian B Lee

Nicholas Claude Mariette

The present document describes a method () for encoding a multi-channel input signal (). The method () comprises determining () a plurality of downmix channel signals () from the multi-channel input signal () and performing () energy compaction of the plurality of downmix channel signals () to provide a plurality of compacted channel signals (). Furthermore, the method () comprises determining () joint coding metadata () based on the plurality of compacted channel signals () and based on the multi-channel input signal (), wherein the joint coding metadata () is such that it allows upmixing of the plurality of compacted channel signals () to an approximation of the multi-channel input signal (). In addition, the method () comprises encoding () the plurality of compacted channel signals () and the joint coding metadata ().

Methods and devices for encoding and/or decoding immersive audio signals

A multi-input, multi-output audio process is implemented as a linear system for use in an audio filterbank to convert a set of frequency-domain input audio signals into a set of frequency-domain output signals. A transfer function from one input to one output is defined as a frequency dependent gain function. In some implementations, the transfer function includes a direct component that is substantially defined as a frequency dependent gain, and one or more decorrelated components that have frequency-varying group phase response. The transfer function is formed from a set of sub-band functions, with each sub-band function being formed from a set of corresponding component transfer functions including direct component and one or more decorrelated components.

Audio filterbank with decorrelating components

A computer implemented system for rendering captured audio soundfields to a listener comprises apparatus to deliver the audio soundfields to the listener. The delivery apparatus delivers the audio soundfields to the listener with first and second audio elements perceived by the listener as emanating from first and second virtual source locations, respectively, and with the first audio element and/or the second audio element delivered to the listener from a third virtual source location. The first virtual source location and the second virtual source location are perceived by the listener as being located to the front of the listener, and the third virtual source location is located to the rear or the side of the listener.

Method of rendering one or more captured audio soundfields to a listener

The disclosure herein generally relates to capturing, acoustic pre-processing, encoding, decoding, and rendering of directional audio of an audio scene. In particular, it relates to a device adapted to modify a directional property of a captured directional audio in response to spatial data of a microphone system capturing the directional audio. The disclosure further relates to a rendering device configured to modify a directional property of a received directional audio in response to received spatial data.

Audio processing in immersive audio services

The positions of a plurality of speakers at a media consumption site are determined. Audio information in an object-based format is received. Gain adjustment value for a sound content portion in the object-based format may be determined based on the position of the sound content portion and the positions of the plurality of speakers. Audio information in a ring-based channel format is received. Gain adjustment value for each ring-based channel in a set of ring-based channels may be determined based on the ring to which the ring-based channel belongs and the positions of the speakers at a media consumption site.

Methods, apparatus and systems for position-based gain adjustment of object-based audio

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.

Processing of microphone signals for spatial playback

The disclosed embodiments enable converting audio signals captured in various formats by various capture devices into a limited number of formats that can be processed by an audio codec (e.g., an Immersive Voice and Audio Services (IVAS) codec). In an embodiment, a simplification unit of the audio device receives an audio signal captured by one or more audio capture devices coupled to the audio device. The simplification unit determines whether the audio signal is in a format that is supported/not supported by an encoding unit of the audio device. Based on the determining, the simplification unit, converts the audio signal into a format that is supported by the encoding unit. In an embodiment, if the simplification unit determines that the audio signal is in a spatial format, the simplification unit can convert the audio signal into a spatial 'mezzanine' format supported by the encoding.

Transforming audio signals captured in different formats into a reduced number of formats for simplifying encoding and decoding operations

Methods and systems for improving signal processing by smoothing the covariance matrix of a multi-channel signal by setting a forgetting factor based on the bins of a band. A method and system for resetting the smoothing based on transient detection is also disclosed. A method and system for resampling for the smoothing during a banding transition is also disclosed.

Systems and methods for covariance smoothing

The disclosure relates to methods of processing a spatial audio signal for generating a compressed representation of the spatial audio signal. The methods include analyzing the spatial audio signal to determine directions of arrival for one or more audio elements; for at least one frequency subband, determining respective indications of signal power associated with the directions of arrival; generating metadata including direction information that includes indications of the directions of arrival of the audio elements, and energy information that includes respective indications of signal power; generating a channel-based audio signal with a predefined number of channels based on the spatial audio signal; and outputting, as the compressed representation, the channel-based audio signal and the metadata. The disclosure further relates to methods of processing a compressed representation of a spatial audio signal for generating a reconstructed representation of the spatial audio signal, and to corresponding apparatus, programs, and storage media.

Multichannel audio encode and decode using directional metadata

Systems, methods, and computer program products of audio processing based on Adaptive Intermediate Spatial Format (AISF) are described. The AISF is an extension to ISF that allows spatial resolution around an ISF ring to be adjusted dynamically with respect to content of incoming audio objects. An AISF encoder device adaptively warps each ISF ring during ISF encoding to adjust angular distance between objects, resulting in increase in uniformity of energy distribution around the ISF ring. At an AISF decoder device, matrices that decode sound positions to the output speaker take into account the warping that was performed at the AISF encoder device to reproduce the true positions of sound sources.

Audio processing in adaptive intermediate spatial format

Some methods involve receiving an input audio signal that includes N input audio channels, the input audio signal representing a first soundfield format having a first soundfield format resolution, N being an integer &#x2265;2. A first decorrelation process may be applied to two or more of the input audio channels to produce a first set of decorrelated channels, the first decorrelation process maintaining an inter-channel correlation of the set of input audio channels. A first modulation process may be applied to the first set of decorrelated channels to produce a first set of decorrelated and modulated output channels. The first set of decorrelated and modulated output channels may be combined with two or more undecorrelated output channels to produce an output audio signal that includes O output audio channels representing a second and relatively higher-resolution soundfield format than the first soundfield format, O being an integer &#x2265;3.

Enhancement of spatial audio signals by modulated decorrelation

Growing community of inventors

Rose Bay, Australia

David S McGrath