Qualcomm Incorporated

Andre Gustavo P Schevciw

Erik Visser

Babak Forutanpour

Brian Lee Momeyer

Jongwon Shin

Sang-Uk Ryu

Lae-Hoon Kim

Nils Günther Peters

Samir Kumar Gupta

Pei Xiang

Kwokleung Chan

Louis Dominic Oliveira

Yinyi Guo

Ren Li

Hyun Jin Park

Wade Lyle Heimbigner

Jeremy Patrick Toman

Hui-Ya Liao Nelson

Nils Gunther Peters

Roy Benjamin Silverstein

Kevin M Beckwith

Selena M Salazar

In general, techniques are described by which to support scalable unified audio rendering. A device comprising an audio decoder, a memory, and a processor may be configured to perform various aspects of the techniques. The audio decoder may decode, from a bitstream, first audio data and second audio data. The memory may store the first audio data and the second audio data. The processor may render the first audio data into first spatial domain audio data for playback by virtual speakers at a set of virtual speaker locations, and render the second audio data into second spatial domain audio data for playback by the virtual speakers at the set of virtual speaker locations. The processor may also mix the first spatial domain audio data and the second spatial domain audio data to obtain mixed spatial domain audio data, and convert the mixed spatial domain audio data to scene-based audio data.

Scalable unified audio renderer

An apparatus, system and method for forwarding a communication from a communication receiving device to a destination device based upon the identity of the user of the destination device.

Communication management utilizing destination device user presence probability

A method for signal level matching by an electronic device is described. The method includes capturing a plurality of audio signals from a plurality of microphones. The method also includes determining a difference signal based on an inter-microphone subtraction. The difference signal includes multiple harmonics. The method also includes determining whether a harmonicity of the difference signal exceeds a harmonicity threshold. The method also includes preserving the harmonics to determine an envelope. The method further applies the envelope to a noise-suppressed signal.

Systems and methods for audio signal processing

A computing device includes a touch screen display with a plurality of force sensors, each of which provides a signal in response to contact with the touch screen display. Using force signals from the plurality of force sensors, a characteristic of the contact is determined, such as the magnitude of the force, the centroid of force and the shear force. The characteristic of the contact is used to select a command which is processed to control the computing device. For example, the command may be related to manipulating data displayed on the touch screen display, e.g., by adjusting the scroll speed or the quantity of data selected in response to the magnitude of force, or related to an operation of an application on the computing device, such as selecting different focal ranges, producing an alarm, or adjusting the volume of a speaker in response to the magnitude of force.

Force sensing touch screen

A mobile platform includes a microphone array and is capable of implementing beamforming to amplify or suppress audio information from a sound source. The sound source is indicated through a user input, such as pointing the mobile platform in the direction of the sound source or through a touch screen display interface. The mobile platform further includes orientation sensors capable of detecting movement of the mobile platform. When the mobile platform moves with respect to the sound source, the beamforming is adjusted based on the data from the orientation sensors so that beamforming is continuously implemented in the direction of the sound source. The audio information from the sound source may be included or suppressed from a telephone or video-telephony conversation. Images or video from a camera may be likewise controlled based on the data from the orientation sensors.

Variable beamforming with a mobile platform

Methods and apparatus for generating an anti-noise signal and equalizing a reproduced audio signal (e.g., a far-end telephone signal) are described, wherein the generating and the equalizing are both based on information from an acoustic error signal.

Systems, methods, devices, apparatus, and computer program products for audio equalization

An original loudness level of an audio signal is maintained for a mobile device while maintaining sound quality as good as possible and protecting the loudspeaker used in the mobile device. The loudness of an audio (e.g., speech) signal may be maximized while controlling the excursion of the diaphragm of the loudspeaker (in a mobile device) to stay within the allowed range. In an implementation, the peak excursion is predicted (e.g., estimated) using the input signal and an excursion transfer function. The signal may then be modified to limit the excursion and to maximize loudness.

Loudness maximization with constrained loudspeaker excursion

Growing community of inventors

San Diego, CA, United States of America

Andre Gustavo P Schevciw