Dolby Laboratories Licensing Corporation

Dolby International Ab

Lake Technology Limited

Dobly Laboratories Licensing Corporation

Dobly International Ab

Glenn N Dickins

David Gunawan

Xuejing Sun

Richard James Cartwright

Dong Shi

David S McGrath

Heiko Purnhagen

Michael Eckert

Hannes Muesch

Alan Jeffrey Seefeldt

Leif Jonas Samuelsson

Joshua Brandon Lando

Christopher Graham Hines

Paul Francis Holmberg

Kai Lin Li

Shen Huang

Daniel Arteaga

Mark R P Thomas

Lars Villemoes

Mark Stuart Vinton

Gary Spittle

Erwin Goesnar

Timothy Neal

Brendon Costa

Zhiwei Shuang

Leif Sehlstrom

Craig Johnston

Yen-Liang Shue

Janusz Klejsa

Gopal Erinjippurath

Ludovic Christophe Malfait

Adam J Mills

Rowan James Katekar

James Zealey

Peng Yin

Taoran Lu

C Phillip Brown

Harald Mundt

Regunathan Radhakrishnan

Toni Hirvonen

Claus Bauer

John Paul Griffin

Kai Li

Wenyu Jiang

Andrew Peter Reilly

Guilin Ma

Nicholas Luke Appleton

Bin Cheng

Phillip Williams

Adam Richard McKeag

Feng Deng

Andrew Border

Hiroaki Kondo

Carlos Eduardo Medaglia Dyonisio

Huiqun Deng

Poppy Crum

JiaQuan Huo

Andreas Ehret

Hariharan Ganapathy-Kathirvelu

Jean-Francois Catz

Tet Fei Yap

Andy Jaspar

Hadis Nosrati

Mark Richard Paul Thomas

Carlos Medaglia Dyonisio

Paul Holmberg

Anthony Tucker

Raphael Marc Ullmann

Jeremy Pascal Brun

Guodong Li

David P Doney

Simon Aridis-Lang

Aya Itokawa

Lori L Solomon

Matthew A Tullis

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

Methods for echo estimation or echo management (echo suppression or cancellation) on an input audio signal, with at least one of adaptation of a sparse prediction filter set, modification (for example, truncation) of adapted prediction filter impulse responses, generation of a composite impulse response from adapted prediction filter impulse responses, or use of echo estimation and/or echo management resources in a manner determined at least in part by classification of the input audio signal as being (or not being) echo free. Other aspects are systems configured to perform any embodiment of any of the methods.

Echo estimation and management with adaptation of sparse prediction filter set

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

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

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, an apparatus, and logic to post-process raw gains determined by input processing to generate post-processed gains, comprising using one or both of delta gain smoothing and decision-directed gain smoothing. The delta gain smoothing comprises applying a smoothing filter to the raw gain with a smoothing factor that depends on the gain delta: the absolute value of the difference between the raw gain for the current frame and the post-processed gain for a previous frame. The decision-directed gain smoothing comprises converting the raw gain to a signal-to-noise ratio, applying a smoothing filter with a smoothing factor to the signal-to-noise ratio to calculate a smoothed signal-to-noise ratio, and converting the smoothed signal-to-noise ratio to determine the second smoothed gain, with smoothing factor possibly dependent on the gain delta.

Post-processing gains for signal enhancement

An apparatus and method of pre-conditioning audio for machine perception. Machine perception differs from human perception, and different processing parameters are used for machine perception applications (e.g., speech to text processing) as compared to those used for human perception applications (e.g., voice communications). These different parameters may result in pre-conditioned audio that is worsened for human perception yet improved for machine perception.

Pre-conditioning audio for echo cancellation in machine perception

Systems and methods for providing forward error correction for a multi-channel audio signal are described. Blocks of an audio stream are buffered into a frame. A transformation can be applied that compacts the energy of each block into a plurality of transformed channels. The energy compaction transform may compact the most energy of a block into the first transformed channel and to compact decreasing amounts of energy into each subsequent transformed channel. The transformed frame may be encoded using any suitable codec and transmitted in a packet over a network. Improved forward error correction may be provided by attaching a low bit rate encoding of the first transformed channel to a subsequent packet. To reconstruct a lost packet, the low bit rate encoding of the first channel for the lost packet may be combined with a packet loss concealment version of the other channels, constructed from a previously-received packet.

Selective forward error correction for spatial audio codecs

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

A rendering mode may be determined for received audio data, including audio signals and associated spatial data. The audio data may be rendered for reproduction via a set of loudspeakers of an environment according to the rendering mode, to produce rendered audio signals. Rendering the audio data may involve determining relative activation of a set of loudspeakers in an environment. The rendering mode may be variable between a reference spatial mode and one or more distributed spatial modes. The reference spatial mode may have an assumed listening position and orientation. In the distributed spatial mode(s), one or more elements of the audio data may each be rendered in a more spatially distributed manner than in the reference spatial mode and spatial locations of remaining elements of the audio data may be warped such that they span a rendering space of the environment more completely than in the reference spatial mode.

Adaptable spatial audio playback

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.

An audio communication endpoint receives a bitstream containing spectral components representing spectral content of an audio signal, wherein the spectral components relate to a first range extending up to a first break frequency, above which any spectral components are unassigned. The endpoint adapts the received bitstream in accordance with a second range extending up to a second break frequency by removing spectral components or adding neutral-valued spectral components relating to a range between the first and second break frequencies. The endpoint then attenuates spectral content in a neighbourhood of the least of the first and second break frequencies for thereby achieving a gradual spectral decay. After this, reconstructing the audio signal is reconstructed by an inverse transform operating on spectral components relating to said second range in the adapted and attenuated received bitstream. At small computational expense, the endpoint may to adapt to different sample rates in received bitstreams.

Enabling sampling rate diversity in a voice communication system

A communication system, method, and computer-readable medium therefor comprise a media server configured to receive a plurality of audio streams from a corresponding plurality of client devices, the media server including circuitry configured to rank the plurality of audio streams based on a predetermined metric, group a first portion of the plurality of audio streams into a first set, the first portion of the plurality of audio streams being the N highest-ranked audio streams, group a second portion of the plurality of audio streams into a second set, the second portion of the plurality of audio streams being the M lowest-ranked audio streams, forward respective audio streams of the first set to a receiver device, and discard respective audio streams of the second set, wherein N and M are independent integers.

Scalable voice scene media server

Growing community of inventors

Como, Australia

Glenn N Dickins