Kopin Corporation

Solos Technology Limited

Other

Dashen Fan

Joseph Yong Kwon

Xi Chelsea Chen

Hua Bao

Eric Frederic Davis

Christopher Iain Parkinson

John C C Fan

Frederick P Herrmann

Xi Chen

Yong Seok Seo

Jang Ho Kim

John Gassel

Murshed Khandaker

Systems and methods are described to extract desired audio from an apparatus to be worn on a user's head. An apparatus includes a head wearable device and an array of at least three microphones. The at least three microphones are arranged along a plurality of at least two non-parallel axes. Selection logic is configured to identify a selected axis from the plurality and two microphones from the array that form the selected axis. A beamformer is configured to accept as inputs, signals from the two microphones and to output a main microphone channel and a reference microphone channel.

Microphone configurations for eyewear devices, systems, apparatuses, and methods

Systems and methods are described to extract desired audio from an apparatus to be worn on a user's wrist. The apparatus includes a wrist wearable device, configured to be worn on the user's wrist. The wrist wearable device includes a first microphone. The first microphone has a first response pattern. The first microphone is coupled to the wrist wearable device. The first microphone is positioned on the wrist wearable device to receive a voice signal from a user when the wrist wearable device is on the user's wrist.

Wrist wearable apparatuses and methods with desired signal extraction

An apparatus to time delay a digital, signal output from an oversampled sensor includes a first time delay element and a second time delay element. The first time delay element has a first input and a first output. The first time delay element is configured to output a time delayed signal that is time delayed by an integer number of sampling clock cycles. An output of the oversampled sensor is connected to the first input of the first time delay element. The second time delay element has a second input and a second output and is configured to output a time delayed signal that is time delayed by an integer number of sampling clock cycles. The first output of the first time delay element is connected to the second input of the second time delay element. A multiplexer has a control input and a multiplexer output. The first output of the first time delay element is connected to a first multiplexer input. The second output of the second time delay element is connected to a second multiplexer input. In operation, time delay information is used to provide a signal to the control input to select a particular multiplexer input for output on the multiplexer output. The output of the oversampled sensor is time delayed by an amount provided by the particular multiplexer input.

Fractional time delay structures in digitally oversampled microphone systems, circuits, and methods

Systems, apparatuses, and methods are described to increase a signal-to-noise ratio difference between a main channel and reference channel. The increased signal-to-noise ratio difference is accomplished with an adaptive threshold for a desired voice activity detector (DVAD) and shaping filters. The DVAD includes averaging an output signal of a reference microphone channel to provide an estimated average background noise level. A threshold value is selected from a plurality of threshold values based on the estimated average background noise level. The threshold value is used to detect desired voice activity on a main microphone channel.

Apparatuses and methods for enhanced speech recognition in variable environments

Programming time delay data in an oversampled sensor includes determining whether to enter Programming Mode based on a value of a system parameter received by the oversampled sensor. Programming Mode is entered when the value of the system parameter corresponds to Programming Mode. The time delay data is programmed in the oversampled sensor during Programming Mode. The oversampled sensor uses the time delay data to time delay its output in an oversampled domain. Programming Mode is exited after a predetermined time has expired relative to when Programming Mode was entered. The system parameter can be a frequency of a sampling clock signal.

Programmable fractional time delay in digitally oversampled microphone systems, circuits, and methods

Systems and methods are described to time delay a signal output from an analog-to-digital converter (ADC). The ADC includes a digital sensor responsive to an analog field quantity. The digital sensor is configured to output an oversampled digital output signal at a sampling clock frequency. A time delay element is configured to receive as an input, the oversampled digital output signal and to output a time delayed oversampled digital output signal. A filter is configured to receive as an input the delayed oversampled digital output signal. The filter low pass filters and decimates to a lower sample rate the delayed oversampled digital output signal. An output includes a low pass filtered decimated delayed digital output signal, where the lower sample rate is less than the sampling clock frequency.

Beamforming using fractional time delay in digitally oversampled sensor systems, apparatuses, and methods

Example embodiments include a method of reducing noise include forming a main signal and one or more reference signals at a beam-former based on at least two received audio signals, detecting voice activity at a voice activity detector, where the voice activity detector receives the main and reference signals and outputting a desired voice activity signal, adaptively cancelling noise at an adaptive noise canceller, where the adaptive noise canceller receives the main, reference, and desired voice activity signals and outputs an adaptive noise cancellation signal, and reducing noise at a noise reducer receiving the desired voice activity and adaptive noise cancellation signals and outputting a desired speech signal.

Noise cancelling microphone apparatus

Systems and apparatuses are described to automatically balance acoustic channel sensitivity. A computer readable medium containing executable computer program instructions, which when executed by an acoustic signal processing system, cause the acoustic signal processing system to perform a process that includes calculating a long-term power level of a main acoustic signal to obtain an averaged main acoustic signal. Segments of the main acoustic signal are excluded from the averaged main acoustic signal using a desired voice activity detection signal. A long-term power level of a reference acoustic signal is calculated to obtain an averaged reference acoustic signal. Segments of the reference acoustic signal are excluded from the averaged reference acoustic signal using the desired voice activity detection signal. An amplitude correction signal is created using the averaged main acoustic signal and the averaged reference acoustic signal. The amplitude correction signal is used to adjust either the main acoustic signal or the reference acoustic signal to automatically balance acoustic channel sensitivity.

Apparatuses and systems for acoustic channel auto-balancing during multi-channel signal extraction

Systems and methods are described to extract desired audio from an apparatus that is worn on a user's head. The apparatus includes a head wearable device. A first microphone is positioned on the head wearable device to receive a voice signal from the user A first signal from the first microphone is input as a main channel to a noise cancellation unit. A second microphone is coupled to the head wearable device. A first acoustic distance between the first microphone and the user's mouth is less than a second acoustic distance between the second microphone and the user's mouth when the head wearable device is on the user's head. A second signal from the second microphone is input as a reference channel to the noise cancellation unit. A first signal-to-noise ratio of the first signal is larger than a second signal-to-noise ratio of the second signal.

Head wearable acoustic system with noise canceling microphone geometry apparatuses and methods

Time delay in digitally oversampled sensor systems, apparatuses, and methods

The invention is a multi-microphone voice processing SoC primarily for head worn applications. It bypasses the use of conventional pre-amp voice CODEC (ADC/DAC) chips all together by replacing their functionality with digital MEMS microphone(s) and digital speaker driver (DSD). Functionality necessary for speech recognition such as noise/echo cancellation, speech compression, speech feature extraction and lossless speech transmission are also integrated into the SoC. One embodiment is a noise cancellation chip for wired, battery powered headsets and earphones, as smart-phone accessory. Another embodiment is as a wireless Bluetooth noise cancellation companion chip. The invention can be used in headwear, eyewear glass, mobile wearable computing, heavy duty military, aviation and industrial headsets and other speech recognition applications in noisy environments.

Digital voice processing method and system for headset computer

A method of interfacing with a serving device from a wearable device worn by a user, the method includes establishing a lossless and wireless data link between the serving device and the wearable device. The method further includes collecting, by the wearable device, audio data from one or more microphones of the wearable device. The method also includes sending, by the wearable device, the collected audio data to the serving device through the lossless and wireless data link. The method may further include providing, by the serving device, speech recognition services associated with the audio data.

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Bellevue, WA, United States of America

Dashen Fan