Qualcomm Incorporated

Digimarc Corporation

Shankar Thagadur Shivappa

Ravi K Sharma

Brett Alan Bradley

S M Akramus Salehin

Dipanjan Sen

Tony F Rodriguez

Nils Günther Peters

Aparna R Gurijala

Moo Young Kim

Ajith Mulki Kamath

Bruce L Davis

Ferdinando Olivieri

Erik Visser

Lae-Hoon Kim

Nils Gunther Peters

Hugh L Brunk

Vojtech Holub

Yang Bai

Bijan Forutanpour

Osama M Alattar

Shuhua Zhang

Robert G Lyons

Scott M Long

Sanghyun Chi

Phi Hung Nguyen

David A Cushman

Tomas Filler

John Douglas Lord

Vasudev Nayak

William Y Conwell

Martin James Morrell

Siddhartha Goutham Swaminathan

Isaac Garcia Munoz

Kurt M Eaton

Jason Filos

Reid Westburg

Richard Paul Walters

Hannes Pessentheiner

Reid Matthew Westburg

Richard F Gibson

Aparna Gurijala

Martin Morrell

Gang Chen

Akramus S M Salehin

In general, techniques are described by which to synchronize enhanced audio transports with backward compatible audio transports. A device comprising a memory and one or more processors may be configured to perform the techniques. The memory may store a backward compatible bitstream conforming to a legacy transport format. The processor may obtain, from the backward compatible bitstream, a first audio transport stream, and obtain, from the backward compatible bitstream, a second audio transport stream. The processor(s) may also obtain, from the backward compatible bitstream, indications representative of synchronization information for the first audio transport stream and the second audio transport stream. The processor(s) may synchronize, based on the indications, the first audio transport stream and the second audio transport to obtain synchronized audio data stream. The processor(s) may obtain, based the synchronized audio data, enhanced audio data, and output the enhanced audio data to one or more speakers.

Synchronizing enhanced audio transports with backward compatible audio transports

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

Audio signal processing enhances audio watermark embedding and detecting processes. Audio signal processes include audio classification and adapting watermark embedding and detecting based on classification. Advances in audio watermark design include adaptive watermark signal structure data protocols, perceptual models, and insertion methods. Perceptual and robustness evaluation is integrated into audio watermark embedding to optimize audio quality relative the original signal, and to optimize robustness or data capacity. These methods are applied to audio segments in audio embedder and detector configurations to support real time operation. Feature extraction and matching are also used to adapt audio watermark embedding and detecting.

Multi-mode audio recognition and auxiliary data encoding and decoding

A device includes one or more processors configured to, during a call, receive a sequence of audio frames from a first device. The one or more processors are configured to, in response to determining that no audio frame of the sequence has been received for a threshold duration since a last received audio frame of the sequence, initiate transmission of a frame loss indication to the first device. The one or more processors are also configured to, responsive to the frame loss indication, receive a set of audio frames of the sequence and an indication of a second playback speed from the first device. The one or more processors are configured to initiate playback, via a speaker, of the set of audio frames based on the second playback speed. The second playback speed is greater than a first playback speed of a first set of audio frames of the sequence.

Call audio playback speed adjustment

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 instructions. The device also includes a processor configured to execute the instructions to obtain spatial audio data representing audio from one or more sound sources. The processor is also configured to execute the instructions to generate first directional audio data based on the spatial audio data. The first directional audio data corresponds to a first arrangement of the one or more sound sources relative to an audio output device. The processor is further configured to generate second directional audio data based on the spatial audio data. The second directional audio data corresponds to a second arrangement of the one or more sound sources relative to the audio output device. The second arrangement is distinct from the first arrangement. The processor is also configured to generate an output stream based on the first directional audio data and the second directional audio data.

Directional audio generation with multiple arrangements of sound sources

The techniques disclosed herein include a first device including one or more processors configured to detect a selection of at least one target object external to the first device, and initiate a channel of communication between the first device and a second device associated with the at least one target object external to the first device. The one or more processors may be configured to receive audio packets, from the second device, in response to the selection of at least one target object external to the device, decode the audio packets, received from the second device, to generate an audio signal. The one or more processors may be configured to output the audio signal based on the selection of the at least one target object external to the first device. The first device includes a memory, coupled to the one or more processors, configured to store the audio packets.

Rendering of sounds associated with selected target objects external to a device

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

Mobile device positioning employs various forms of audio signal structures and detection methodologies. In one method, detection of an audio signal from a first source enables construction of a signal to facilitate detection of an audio signal from another source. Signals detected from these sources enable positioning of the mobile device receiving those signals. Another method forms audio signals transmitted from audio sources so that they have parts that add constructively and parts that differentiate the sources to enable positioning. Another audio signal based positioning method adaptively switches among positioning methods so that positioning remains operative as a mobile device moves toward and away from the sources. Another method tracks positioning history, evaluates it for errors and performs error mitigation to improve accuracy. Various other positioning technologies are detailed as well.

Mobile device indoor navigation

A device for communication includes one or more processors configured to receive, during an online meeting, a speech audio stream representing speech of a first user. The one or more processors are also configured to receive a text stream representing the speech of the first user. The one or more processors are further configured to selectively generate an output based on the text stream in response to an interruption in the speech audio stream.

Systems and methods of handling speech audio stream interruptions

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

The techniques disclosed herein include a first device for reading one or more tags in metadata, the first device including one or more processors configured to receive metadata, from a second device, wirelessly connected via a sidelink channel to the first device. The one or more processors may also be configured to read the metadata, received from the second device to extract one or more tags representative of audio content, and identify audio content based on the one or more tags, and output the audio content. The first device may also include a memory, coupled to the one or more processors, configured to store the metadata.

Media exchange between devices

In general, techniques are described by which to provide priority information for higher order ambisonic (HOA) audio data. A device comprising a memory and a processor may perform the techniques. The memory stores HOA coefficients of the HOA audio data, the HOA coefficients representative of a soundfield. The processor may decompose the HOA coefficients into a sound component and a corresponding spatial component, the corresponding spatial component defining shape, width, and directions of the sound component, and the corresponding spatial component defined in a spherical harmonic domain. The processor may also determine, based on one or more of the sound component and the corresponding spatial component, priority information indicative of a priority of the sound component relative to other sound components of the soundfield, and specify, in a data object representative of a compressed version of the HOA audio data, the sound component and the priority information.

Higher order ambisonic audio data

Growing community of inventors

San Diego, CA, United States of America

Shankar Thagadur Shivappa