Dolby Laboratories Licensing Corporation

Dolby International Ab

Microsoft Technology Licensing, LLC

Mark R P Thomas

Daniel Arteaga

Alan Jeffrey Seefeldt

Christopher Graham Hines

Glenn N Dickins

David Gunawan

Richard James Cartwright

Ivan Jelev Tashev

Joshua Brandon Lando

David Emerson Johnston

John Carlton Platt

Nicolas R Tsingos

Christof Fersch

Davide Scaini

Benjamin John Southwell

Olha Michelle Townsend

Piotr Tadeusz Bilinski

Jens Ahrens

Avery Bruni

Heiko Purnhagen

Kristofer Kjoerling

Sven Kordon

Alexander Krueger

David S McGrath

C Phillip Brown

Charles Quito Robinson

Michael John Smithers

Johann-Markus Batke

Hannes Gamper

Carlos Eduardo Medaglia Dyonisio

Archontis Politis

Rowan James Katekar

Jan-Hendrik Hanschke

Carlos Medaglia Dyonisio

Richard J Cartwright

Jeremiha Emile Douglas

David S Mcgrath

A method may involve: receiving direction of arrival (DOA) data corresponding to sound emitted by at least a first smart audio device of the audio environment that includes a first audio transmitter and a first audio receiver, the DOA data corresponding to sound received by at least a second smart audio device of the audio environment that includes a second audio transmitter and a second audio receiver, the DOA data corresponding to sound emitted by at least the second smart audio device and received by at least the first smart audio device; receiving one or more configuration parameters corresponding to the audio environment, to one or more audio devices, or both; and minimizing a cost function based at least in part on the DOA data and the configuration parameter(s), to estimate a position and an orientation of at least the first smart audio device and the second smart audio device.

Automatic localization of audio devices

Some methods may involve receiving a first content stream that includes first audio signals, rendering the first audio signals to produce first audio playback signals, generating first calibration signals, generating first modified audio playback signals by inserting the first calibration signals into the first audio playback signals, and causing a loudspeaker system to play back the first modified audio playback signals, to generate first audio device playback sound. The method(s) may involve receiving microphone signals corresponding to at least the first audio device playback sound and to second through Nth audio device playback sound corresponding to second through Nth modified audio playback signals (including second through Nth calibration signals) played back by second through Nth audio devices, extracting second through Nth calibration signals from the microphone signals and estimating at least one acoustic scene metric based, at least partly, on the second through Nth calibration signals.

Pervasive acoustic mapping

Some methods involve causing a plurality of audio devices in an audio environment to reproduce audio data, each audio device of the plurality of audio devices including at least one loudspeaker and at least one microphone, determining audio device location data including an audio device location for each audio device of the plurality of audio devices and obtaining microphone data from each audio device of the plurality of audio devices. Some methods involve determining a mutual audibility for each audio device of the plurality of audio devices relative to each other audio device of the plurality of audio devices, determining a user location of a person in the audio environment, determining a user location audibility of each audio device of the plurality of audio devices at the user location and controlling one or more aspects of audio device playback based, at least in part, on the user location audibility.

Audibility at user location through mutual device audibility

A method for selecting a device for audio processing may involve receiving a first wakeword confidence metric from a first device that includes at least a first microphone and receiving a second wakeword confidence metric from a second device that includes at least a second microphone. The first and second wakeword confidence metrics may correspond to a first local maximum of a first plurality of wakeword confidence values determined by the first device and a second local maximum of a second plurality of wakeword confidence values determined by the second device. The method may involve comparing the first wakeword confidence metric and the second wakeword confidence metric and selecting a device for subsequent audio processing based, at least in part, on a comparison of the first wakeword confidence metric and the second wakeword confidence metric.

Multi-device wakeword detection

An audio session management method for an audio environment having multiple audio devices may involve receiving, from a first device implementing a first application and by a device implementing an audio session manager, a first route initiation request to initiate a first route for a first audio session. The first route initiation request may indicate a first audio source and a first audio environment destination. The first audio environment destination may correspond with at least a first person in the audio environment, but in some instances will not indicate an audio device. The method may involve establishing a first route corresponding to the first route initiation request. Establishing the first route may involve determining a first location of at least the first person in the audio environment, determining at least one audio device for a first stage of the first audio session and initiating or scheduling the first audio session.

Coordination of audio devices

A method for estimating an audio device location in an environment may involve obtaining direction of arrival (DOA) data for each audio device of a plurality of audio devices in the environment and determining interior angles for each of a plurality of triangles based on the DOA data. Each triangle may have vertices that correspond with audio device locations. The method may involve determining a side length for each side of each of the triangles, performing a forward alignment process of aligning each of the plurality of triangles produce a forward alignment matrix and performing a reverse alignment process of aligning each of the plurality of triangles in a reverse sequence to produce a reverse alignment matrix. A final estimate of each audio device location may be based, at least in part, on values of the forward alignment matrix and values of the reverse alignment matrix.

Audio device auto-location

Some disclosed methods involve encoding or decoding directional audio data. Some encoding methods may involve receiving a mono audio signal corresponding to an audio object and a representation of a radiation pattern corresponding to the audio object. The radiation pattern may include sound levels corresponding to plurality of sample times, a plurality of frequency bands and a plurality of directions. The methods may involve encoding the mono audio signal and encoding the source radiation pattern to determine radiation pattern metadata. Encoding the radiation pattern may involve determining a spherical harmonic transform of the representation of the radiation pattern and compressing the spherical harmonic transform to obtain encoded radiation pattern metadata.

Methods, apparatus and systems for encoding and decoding of directional sound sources

Some methods may involve receiving a first content stream that includes first audio signals, rendering the first audio signals to produce first audio playback signals, generating first direct sequence spread spectrum (DSSS) signals, generating first modified audio playback signals by inserting the first DSSS signals into the first audio playback signals, and causing a loudspeaker system to play back the first modified audio playback signals, to generate first audio device playback sound. The method(s) may involve receiving microphone signals corresponding to at least the first audio device playback sound and to second through Naudio device playback sound corresponding to second through Nmodified audio playback signals (including second through NDSSS signals) played back by second through Nth audio devices, extracting second through NDSSS signals from the microphone signals and estimating at least one acoustic scene metric based, at least partly, on the second through NDSSS signals.

Orchestration of acoustic direct sequence spread spectrum signals for estimation of acoustic scene metrics

A multi-stream rendering system and method may render and play simultaneously a plurality of audio program streams over a plurality of arbitrarily placed loudspeakers. At least one of the program streams may be a spatial mix. The rendering of said spatial mix may be dynamically modified as a function of the simultaneous rendering of one or more additional program streams. The rendering of one or more additional program streams may be dynamically modified as a function of the simultaneous rendering of the spatial mix.

Managing playback of multiple streams of audio over multiple speakers

A method for estimating a user's location in an environment may involve receiving output signals from each microphone of a plurality of microphones in the environment. At least two microphones of the plurality of microphones may be included in separate devices at separate locations in the environment and the output signals may correspond to a current utterance of a user. The method may involve determining multiple current acoustic features from the output signals of each microphone and applying a classifier to the multiple current acoustic features. Applying the classifier may involve applying a model trained on previously-determined acoustic features derived from a plurality of previous utterances made by the user in a plurality of user zones in the environment. The method may involve determining, based at least in part on output from the classifier, an estimate of the user zone in which the user is currently located.

Acoustic zoning with distributed microphones

An audio processing method may involve receiving output signals from each microphone of a plurality of microphones in an audio environment, the output signals corresponding to a current utterance of a person and determining, based on the output signals, one or more aspects of context information relating to the person, including an estimated current proximity of the person to one or more microphone locations. The method may involve selecting two or more loudspeaker-equipped audio devices based, at least in part, on the one or more aspects of the context information, determining one or more types of audio processing changes to apply to audio data being rendered to loudspeaker feed signals for the audio devices and causing one or more types of audio processing changes to be applied. In some examples, the audio processing changes have the effect of increasing a speech to echo ratio at one or more microphones.

Acoustic echo cancellation control for distributed audio devices

An audio session management method may involve: determining, by an audio session manager, one or more first media engine capabilities of a first media engine of a first smart audio device, the first media engine being configured for managing one or more audio media streams received by the first smart audio device and for performing first smart audio device signal processing for the one or more audio media streams according to a first media engine sample clock; receiving, by the audio session manager and via a first application communication link, first application control signals from the first application; and controlling the first smart audio device according to the first media engine capabilities, by the audio session manager, via first audio session management control signals transmitted to the first smart audio device via a first smart audio device communication link and without reference to the first media engine sample clock.

Some disclosed methods involve multi-band bass management. Some such examples may involve applying multiple high-pass and low-pass filter frequencies for the purpose of bass input management. Some disclosed methods treat at least some low-frequency signals as audio objects that can be panned. Some disclosed methods involve panning low and high frequencies separately. Following high-pass rendering, a power audit may determine a low-frequency deficit factor that is to be reproduced by subwoofers or other low-frequency-capable loudspeakers.

Methods and devices for bass management

The present document describes a method () for encoding a soundfield representation (SR) input signal () describing a soundfield at a reference position, wherein the SR input signal () comprises a plurality of channels for a plurality of different directivity patterns of the soundfield at the reference position. The method () comprises extracting () one or more audio objects () from the SR input signal (). Furthermore, the method () comprises determining () a residual signal () based on the SR input signal () and based on the one or more audio objects (). The method () also comprises performing joint coding of the one or more audio objects () and/or the residual signal (). In addition, the method () comprises generating () a bitstream () based on data generated in the context of joint coding of the one or more audio objects () and/or the residual signal ().

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Bellevue, WA, United States of America

Mark R P Thomas