Systems and methods are disclosed in which a playback device transmits a first sound signal including a predetermined waveform. In one example, the playback device receives a second sound signal including at least one reflection of the first sound signal. The second sound signal is processed to determine a location of a person relative to the playback device, and a characteristic of audio reproduction by the playback device is selected, based on the determined location of the person.

A system for monitoring a building having one or more microphones coupled to a telephone includes a detector configured to detect a triggering event within the building and transmit an activating signal when the triggering event is detected, and a control module configured to receive the activating signal from the detector. The control module is programmed to activate at least one of the one or more microphones to monitor sound when the activating signal is received.

The present disclosure relates generally to improving audio processing using an intelligent microphone and, more particularly, to techniques for processing audio received at a microphone with integrated analog-to-digital conversion, digital signal processing, acoustic source separation, and for further processing by a speech recognition system. Embodiments of the present disclosure include intelligent microphone systems designed to collect and process high-quality audio input efficiently. Systems and method for audio processing using an intelligent microphone include an integrated package with one or more microphones, analog-to-digital converters (ADCs), digital signal processors (DSPs), source separation modules, memory, and automatic speech recognition. Systems and methods are also provided for audio processing using an intelligent microphone that includes a microphone array and uses a preprogrammed audio beamformer calibrated to the included microphone array.

This application relates to methods and apparatus for digital microphones. Disclosed is a digital microphone apparatus (300) for outputting a digital output signal (DATA) at a sample rate defined by a received clock signal (CLK). The apparatus includes a band splitter (302) configured to receive a microphone signal (S.sub.MD) indicative of an output of a microphone transducer and split said microphone signal into first signal path (S.sub.P1) for frequencies in a first band and a second signal path (S.sub.P2) for frequencies in a second band, the frequencies of the second band being higher than the frequencies in the first band. A modulation block (304) is configured to operate on the second signal path to down-convert signals in the second signal path from the second frequency band to a lower frequency band.

Systems, methods, and devices for determining a frequency of a received audio signal. A device may include a microphone, a speaker, and a processor configured to perform operations including receiving, using the microphone, a first audio signal having a predetermined frequency. The operations may also include, for each test frequency of a plurality of test frequencies and for each test phase of a plurality of test phases, generating a second audio signal having the test frequency and the test phase, outputting, using the speaker, the second audio signal, receiving, using the microphone, a combined audio signal being a combination of the first audio signal and the second audio signal, determining an amplitude of the combined audio signal, and determining that the predetermined frequency is within a threshold range of the test frequency when the amplitude of the combined audio signal is below a threshold.

A signal processing method for cochlear implant is performed by a speech processor and comprises a noise reduction stage and a signal compression stage. The noise reduction stage can efficiently reduce noise in a electrical speech signal of a normal speech. The signal compression stage can perform good signal compression to enhance signals to stimulate cochlear nerves of a patient with hearing loss. The patient who uses a cochlear implant performing the signal processing method of the present disclosure can accurately hear normal speech.

According to embodiments, a sensor includes a structure body, a container, a liquid and a sensing unit. The structure body includes a supporter, and a film unit. The film unit includes a first region. The first region includes a first end portion supported by the supporter, and a first portion being displaceable. The film unit includes an opening. The container is connected to the structure body. A first space is defined between the film unit and the container. The liquid is provided inside the first space. The sensing unit senses a displacement of the first portion accompanying a displacement of the liquid.

Laser-based devices utilizing temperature modulation for improved self-mix sensing. A self-mix laser unit includes: an active region having a first side and a second, opposite, side; a p-type Distributed Bragg Reflector (DBR) region, which is in direct touch with said first side of the active region; an n-type DBR region, which is in direct touch with the second side of the active region; and an n-type or p-type or other substrate. A heating unit provides modulated heating to the active region, either directly via an electrical resistor within the active region; or indirectly by passing or propagating modulated heat through one of the DBR regions or through the substrate. The modulated heating improves the laser-based self-mix signal.

Systems, apparatuses and methods for secure audio acquisition. The method includes receiving audio data via a digital microphone. The digital microphone outputs a single bit at a high sampling rate. The digital microphone output is converted to a full range audio signal. The full range audio signal is filtered to provide a band limited audio output that avoids capture of enough of a spectral range of speech for the speech to be intelligible.

The present disclosure provides a transistor acoustic sensor element and a method for manufacturing the same, an acoustic sensor and a portable device. The transistor acoustic sensor element comprises a gate, a gate insulating layer, a first electrode, an active layer and a second electrode arranged on a base substrate, wherein the active layer has a nanowire three-dimensional mesh structure and thus can vibrate under the action of sound signals, so that the output current of the transistor acoustic sensor element changes correspondingly. Since the active layer having the nanowire three-dimensional mesh structure can sensitively sense weak vibration of acoustic waves, the sensitivity to sound signals of the transistor acoustic sensor element is improved.