Audience, Inc.

Knowles Electronics, Inc.

Peter Santos

Carlo Murgia

Michael M Goodwin

Lloyd Watts

Carlos M Avendano

Dana C Massie

Ye Jiang

Jean Laroche

Brian D Clark

David Klein

Umit Yapanel

Sridhar Krishna Nemala

Matthew Cowan

Hong You

Methods for voice sensing and keyword analysis are provided. An example method allows for causing a mobile device to transition to a second power mode, from a first power mode, in response to a first acoustic signal. The method includes authenticating a user based at least in part on a second acoustic signal. While authenticating the user, the second acoustic signal is compared to a spoken keyword. The spoken keyword is analyzed for authentication strength based on the length of the spoken keyword, quality of a series of phonemes used to represent the spoken keyword, and likelihood of the series of phonemes to be detected by a voice sensing. While receiving the first and second acoustic signals, a signal to noise ratio (SNR) is determined. The SNR is used to adjust sensitivity of a detection threshold of a voice sensing.

Voice sensing and keyword analysis

Systems and methods for utilizing inter-microphone level differences to attenuate noise and enhance speech are provided. In exemplary embodiments, energy estimates of acoustic signals received by a primary microphone and a secondary microphone are determined in order to determine an inter-microphone level difference (ILD). This ILD in combination with a noise estimate based only on a primary microphone acoustic signal allow a filter estimate to be derived. In some embodiments, the derived filter estimate may be smoothed. The filter estimate is then applied to the acoustic signal from the primary microphone to generate a speech estimate.

System and method for utilizing inter-microphone level differences for speech enhancement

An audio processing system processes an audio signal that may come from one or more microphones. The audio processing system may use information from one or more non-acoustic sensors to improve a variety of system characteristics, including responsiveness and quality. Those audio processing systems that use spatial information, for example to separate multiple audio sources, are undesirably susceptible to changes in the relative position of any audio sources, the audio processing system itself, or any combination thereof. Using the non-acoustic sensor information may decrease this susceptibility advantageously in an audio processing system.

Selection of system parameters based on non-acoustic sensor information

Provided are systems and methods for dynamic detection of changes in signal bandwidth associated with audio samples received by a communication device during handovers, where the communication device is moved through geographical areas served by different wireless network technologies. Based on the detection, operation parameters of an audio processor of the communication device may be adjusted or switched to achieve optimal performance of audio enhancing procedures (e.g., noise suppression) performed by the audio processor and to preserve minimal power consumption.

Dynamic bandwidth change detection for configuring audio processor

An audio processing system processes an audio signal that may come from one or more microphones. The audio processing system may use information from one or more non-acoustic sensors to improve a variety of system characteristics, including responsiveness and quality. Especially those audio processing systems that use spatial information, for example to separate multiple audio sources, are undesirably susceptible to changes in the relative position of any audio sources, the audio processing system itself, or any combination thereof. Using the non-acoustic sensor information may decrease this susceptibility advantageously in an audio processing system.

Background audio can be provided during telephonic communication. Telephonic communication can be established via a network, such as between a user of a telephony device and a communication partner having a second telephony device. A voice signal may be received from the user via a microphone integral with the telephony device. An audio track can be retrieved, for example, from memory integral with the telephony device or from a third-party service provider via the communications network. Noise reduction is performed on the voice signal to produce a clean voice signal. The clean voice signal may be combined with the audio track to produce a combined signal, such that the audio track provides background audio to the clean voice signal. The combined signal can then be transmitted from the telephony device to the second telephony device via the communications network.

Providing background audio during telephonic communication

Disclosed are methods and devices for communicating between a user device, such as a mobile terminal, and a three wire headset assembly having two microphones coupled thereto. The methods provide two microphone uplink communication and stereo audio playback on a three wire headset assembly without requiring a fourth wire or the addition of a separate voice processor. The headset assembly includes a primary microphone and secondary microphone spaced some distance apart for applying techniques of noise reduction and suppression. The headset assembly includes two speakers. Controlled switches are also included in both the user device and headset assembly to achieve dual use for a single wire of the three wires. The three wire headset may advantageously support both stereo playback and two microphone voice processing. In some embodiments, two microphone noise suppression for a wired headset mode of the user device is provided.

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Los Altos, CA, United States of America

Peter Santos