Dsp Concepts, Inc.

Gopro, Inc.

Sirf Technology, Inc.

Other

Analog Devices,inc.

Paul Eric Beckmann

Erich Tisch

Ke Li

Zhinian Jing

Shashank Bharathi

William Kerry Keal

Erik E Anderson

Andrew Chou

Timothy S Stilson

Julien Basch

Arpit Shah

Arvind Ramanthan

Eric Jing

Jeffrey M Brockmole

A method of processing microphone signals generates a specified or required multichannel microphone input for a voice recognition system. The method comprises receiving first and second microphone signal from first and second microphones that have a spacing that is required by a voice response or recognition system. The first and second microphone signals are processed using at least one additional microphone signal to generate a first modified microphone signal and a second modified microphone signal respectively. The first and second modified microphone signals are then provided as required microphone inputs to a voice response or recognition system.

Processing of microphone signals required by a voice recognition system

A method of providing custom audio processing in an audio device having an embedded audio DSP framework is disclosed. The audio device has an associated application that runs on a personal computing device. The method comprises receiving a notification of the availability of an audio processing configuration file, the audio processing configuration file identifying audio modules, interconnections between the audio modules and related settings for custom audio processing, transmitting a request for the audio processing configuration file from the personal computing device, receiving the audio processing configuration file at the personal computing device, transmitting the audio processing configuration file to the audio device from the personal computing device, and loading the audio processing configuration file into the audio device.

Downloadable audio features

A system and method for improving the performance of a hands-free voice user interface system while minimizing the computational complexity without sacrificing performance. Specifically, when estimating the location of the talker for the purpose of steering a directional beam in the direction of the active talker. A hands-free voice user interface system requires a clean signal to be streamed to the cloud for recognition. One way to improve the speech signal is to estimate where the talker is and steer a beam in the direction of the active talker. To locate the talker to a localized position, a direction of arrival estimator (DOA) algorithm is used. DoA generally requires noise and echo free signal for optimal estimation, but it is computationally expensive to run audio pre-processing such as an acoustic echo cancellation for each microphone in microphone array. To reduce computational complexity, the system and method extract certain range of frequency and operate pre-processing only on the selected frequency. By properly selecting the frequency range, it does not degrade DoA accuracy while significantly reducing computational complexity.

Narrowband direction of arrival for full band beamformer

Generating an audio processing design includes placing representations of audio processing modules using a user interface. The method includes causing the display of the representations of the audio processing modules in the user interface, receiving user input to place a representation of a start thread module in the audio processing design between a representation of an upstream audio module and representations of downstream audio modules, the representation of the start thread module having data associated therewith that defines a buffer up factor and a thread identifier, receiving user input to specify the buffer up factor and thread identifier associated with the representation of the start thread module, and applying the specified thread identifier and a sample block size that is derived from the specified buffer up factor, downstream from the representation of the thread start module.

Audio multithreading interface

A method and apparatus for handling both echo cancellation and noise in a given space using a single adaptive filter. The proposed method utilizes just a single adaptive filter and without any control logic, the system is capable of handling both the AEC and the noise in a given space. The technology as proposed and claimed herein proposes using a single adaptive filter to cancel referenced noise AEC and unreferenced noise ANC in a given physical space using a beamformer. By adding the far end reference and a beam pointed dynamically on the noise source in the given physical space this can be achieved. This invention also reduces the processing load and memory.

Method and system for using single adaptive filter for echo and point noise cancellation

An apparatus and method for an audio user interface solution that avoids the time variation problems seen when using an acoustic echo canceler in an audio system. The solution as disclosed and claimed herein results in lower CPU usage in the audio system because a synchronous sample rate correction (SSRC) is performed on the sample rate multiple rate content and the asynchronous sample rate correction (ASRC) is applied on a small number of signals used to feed the ASRC.

Apparatus and method for audio user interface processing with disparate sampling rates

A camera system includes a first microphone, a second microphone, and a microphone controller. The first microphone and the second microphone are configured to capture audio over a time interval to produce a first captured audio signal and a second captured audio signal, respectively. The second captured audio signal is dampened relative to the first captured audio signal by a dampening factor. The microphone controller is configured to store the first captured audio signal in response to a determination that the first captured audio signal does not clip. In response to a determination that the first captured audio signal clips, the microphone controller is configured to identify a gain between the first captured audio signal and the second captured audio signal representative of the dampening factor, amplify the second captured audio signal based on the identified gain, and store the amplified second captured audio signal.

Audio signal level estimation in cameras

An audio capture system for a sports camera includes at least one 'enhanced' microphone and at least one 'reference' microphone. The enhanced microphone includes a drainage enhancement feature to enable water to drain from the microphone more quickly than the reference microphone. A microphone selection controller selects between the microphones based on a microphone selection algorithm to enable high quality in conditions where the sports camera transitions in and out of water during activities such as surfing, water skiing, swimming, or other wet environments.

Automatic microphone selection in a sports camera

Methods and systems for processing a Global Positioning System (GPS) signal are provided. A method includes: transforming a time domain GPS signal to a frequency domain GPS signal; storing a frequency domain pseudorandom noise (PRN) signal; correlating the frequency domain GPS signal with the PRN signal at a plurality of frequencies, said correlating including: shifting the frequency domain GPS signal by an amount corresponding to one of the plurality of frequencies; downsampling the shifted frequency domain GPS signal; and multiplying the shifted frequency domain GPS signal by the stored frequency domain PRN signal to produce a correlated signal.

Reducing and sharing computations for GPS signal processing

A sampling-rate conversion method receives N input channels which have been digitized at an input sampling rate, and converts the sampling rate of each input channel to produce N output channels at an output sampling rate. The method comprises computing a common FIR interpolating function which depends upon the input and output sample clocks and the instantaneous output time, and applying the common FIR interpolating function to all N input channels to compute all N output channels.

Multi-channel sample rate conversion method

Methods and devices for processing GPS signals are provided. The device includes: a memory for storing one or more lookup tables, each lookup table including a plurality of entries, each entry including an input segment and an output segment, wherein the output segment is a system response to the corresponding input segment. The device further includes a processor configured to receive an input data sequence including one or more input segments, and for each of the input segments, retrieve from one of the lookup tables the output segments corresponding to the input segment. The processor is further configured to perform a time-shifted sum of one or more output segments to produce an output sequence that is a downsampled representation of the input data sequence.

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Sunnyvale, CA, United States of America

Paul Eric Beckmann