10722results about "Microphones" patented technology

An wearable communications garment that includes one or more user-selectable inputs integrated into the garment. A sartorial communications apparatus may include a flexible material that is worn (e.g., as an undergarment) by the user and includes one or more interactive sensors that may be manually activated by a user, even through one or more intervening layers of clothing. The apparatus may also include one or more additional body sensors configured to sense a user's position, movement, and / or physiological status. The sensor(s) may be connected via a conductive trace on the garment to a sensor module for analysis and / or transmission. Methods of manufacturing the garments as well as methods of using the garments are also described.

Apparatus and method that improves speech recognition accuracy, by monitoring the position of a user's headset-mounted speech microphone, and prompting the user to reconfigure the speech microphone's orientation if required. A microprocessor or other application specific integrated circuit provides a mechanism for comparing the relative transit times between a user's voice, a primary speech microphone, and a secondary compliance microphone. The difference in transit times may be used to determine if the speech microphone is placed in an appropriate proximity to the user's mouth. If required, the user is automatically prompted to reposition the speech microphone.

A power delivery method and system for powering a headset. A power signal is combined with an audio signal to form a composite signal that is communicated over a shared channel to the headset. The power signal is generated by modulating a carrier signal with a modulation signal. The modulation signal is derived from the amplitude of the audio signal so that the peak levels of the composite signal do not exceed the maximum allowable output of an audio I / O circuit driving the headset.

A communication component modifies production of an audio waveform at determined modification segments to thereby mitigate the effects of a delay in processing and / or receiving a subsequent audio waveform. The audio waveform and / or data associated with the audio waveform are analyzed to identify the modification segments based on characteristics of the audio waveform and / or data associated therewith. The modification segments show where the production of the audio waveform may be modified without substantially affecting the clarity of the sound or audio. In one embodiment, the invention modifies the sound production at the identified modification segments to extend production time and thereby mitigate the effects of delay in receiving and / or processing a subsequent audio waveform for production.

Sound processing methods and apparatus are provided. A sound capture unit is configured to identify one or more sound sources. The sound capture unit generates data capable of being analyzed to determine a listening zone at which to process sound to the substantial exclusion of sounds outside the listening zone. Sound captured and processed for the listening zone may be used for interactivity with the computer program. The listening zone may be adjusted based on the location of a sound source. One or more listening zones may be pre-calibrated. The apparatus may optionally include an image capture unit configured to capture one or more image frames. The listening zone may be adjusted based on the image. A video game unit may be controlled by generating inertial, optical and / or acoustic signals with a controller and tracking a position and / or orientation of the controller using the inertial, acoustic and / or optical signal.

Described herein are systems and methods for providing media playback in a networked environment. In one embodiment, a networked media playback device is configured to provide a web server for delivering data indicative of a browser-renderable control interface for the networked media playback device, or for a related networked media playback system. In overview, the user of a networked device, such as a wireless web enabled device, is able to render the control interface in a web browser, and in this manner control playback of digital media via the networked media playback device or networked media playback system.

A personal sound system is described that includes a wireless network supporting an ear-level module, a companion module and a phone. Other audio sources are supported as well. A configuration processor configures the ear-level module and the companion module for private communications, and configures the ear-level module for a plurality of signal processing modes, including a hearing aid mode, for a corresponding plurality of sources of audio data. The ear module is configured to handle variant audio sources, and control switching among them.

A wireless headset system for use with, e.g. mobile phones, and which incorporates a wireless headset which communicates with a base station via magnetic inductive coupling or radio frequency signals to dial and send or receive calls via a conventional phone, e.g. a mobile phone, attached to the base station. In one embodiment, the mobile phone is carried in the base station which is worn by the user.

A method for improving the quality of a speech signal extracted from a noisy acoustic environment is provided. In one approach, a signal separation process is associated with a voice activity detector. The voice activity detector is a two-channel detector, which enables a particularly robust and accurate detection of voice activity. When speech is detected, the voice activity detector generates a control signal. The control signal is used to activate, adjust, or control signal separation processes or post-processing operations to improve the quality of the resulting speech signal. In another approach, a signal separation process is provided as a learning stage and an output stage. The learning stage aggressively adjusts to current acoustic conditions, and passes coefficients to the output stage. The output stage adapts more slowly, and generates a speech-content signal and a noise dominant signal. When the learning stage becomes unstable, only the learning stage is reset, allowing the output stage to continue outputting a high quality speech signal.

A detector located on or near an ear piece may automatically detect whether a left or right ear or neither ear is wearing the ear piece. A signal mixer may automatically apply a correct configuration to signals transmitted to or from the ear piece according to whether the left or right ear or neither ear is determined to be wearing the ear piece. It is emphasized that this abstract is provided to comply with the rules requiring an abstract that will allow a searcher or other reader to quickly ascertain the subject matter of the technical disclosure. It is submitted with the understanding that it will not be used to interpret or limit the scope or meaning of the claims.

A silicon condenser microphone package is disclosed. The silicon condenser microphone package comprises a transducer unit, substrate, and a cover. The substrate includes an upper surface having a recess formed therein. The transducer unit is attached to the upper surface of the substrate and overlaps at least a portion of the recess wherein a back volume of the transducer unit is formed between the transducer unit and the substrate. The cover is placed over the transducer unit and includes an aperture.

The invention includes a method, apparatus, and computer program to selectively suppress wind noise while preserving narrow-band signals in acoustic data. Sound from one or several microphones is digitized into binary data. A time-frequency transform is applied to the data to produce a series of spectra. The spectra are analyzed to detect the presence of wind noise and narrow band signals. Wind noise is selectively suppressed while preserving the narrow band signals. The narrow band signal is interpolated through the times and frequencies when it is masked by the wind noise. A time series is then synthesized from the signal spectral estimate that can be listened to. This invention overcomes prior art limitations that require more than one microphone and an independent measurement of wind speed. Its application results in good-quality speech from data severely degraded by wind noise.

An energy harvesting device includes a MEMS composite transducer. The MEMS composite transducer includes a substrate. Portions of the substrate define an outer boundary of a cavity. A MEMS transducing member includes a beam having a first end and a second end. The first end is anchored to the substrate and the second end cantilevers over the cavity. A compliant membrane is positioned in contact with the MEMS transducing member. A first portion of the compliant membrane covers the MEMS transducing member. A second portion of the compliant membrane is anchored to the substrate. The compliant member is configured to be set into oscillation by excitations produced externally relative to the energy harvesting device.

Detachable wearable electronic eyeglasses and head mounted gear with a plurality of electronic functions and interchangeable electronic function, and a wearable computer with optimal weight distribution and stretchable arms.

Targeted sound detection methods and apparatus are disclosed. A microphone array has two or more microphones M0 . . . MM. Each microphone is coupled to a plurality of filters. The filters are configured to filter input signals corresponding to sounds detected by the microphones thereby generating a filtered output. One or more sets of filter parameters for the plurality of filters are pre-calibrated to determine one or more corresponding pre-calibrated listening zones. Each set of filter parameters is selected to detect portions of the input signals corresponding to sounds originating within a given listening zone and filter out sounds originating outside the given listening zone. A particular pre-calibrated listening zone is selected at a runtime by applying to the plurality of filters a set of filter coefficients corresponding to the particular pre-calibrated listening zone. As a result, the microphone array may detect sounds originating within the particular listening sector and filter out sounds originating outside the particular listening zone. Sounds are detected with the microphone array. A particular listening zone containing a source of the sound is identified. The sound or the source of the sound is characterized and the sound is emphasized or filtered out depending on how the sound is characterized.

In one embodiment, a directional microphone array having (at least) two microphones generates forward and backward cardioid signals from two (e.g., omnidirectional) microphone signals. An adaptation factor is applied to the backward cardioid signal, and the resulting adjusted backward cardioid signal is subtracted from the forward cardioid signal to generate a (first-order) output audio signal corresponding to a beampattern having no nulls for negative values of the adaptation factor. After low-pass filtering, spatial noise suppression can be applied to the output audio signal. Microphone arrays having one (or more) additional microphones can be designed to generate second- (or higher-) order output audio signals.

Spatial noise suppression for audio signals involves generating a ratio of powers of difference and sum signals of audio signals from two microphones and then performing noise suppression processing, e.g., on the sum signal where the suppression is limited based on the power ratio. In certain embodiments, at least one of the signal powers is filtered (e.g., the sum signal power is equalized) prior to generating the power ratio. In a subband implementation, sum and difference signal powers and corresponding the power ratio are generated for different audio signal subbands, and the noise suppression processing is performed independently for each different subband based on the corresponding subband power ratio, where the amount of suppression is derived independently for each subband from the corresponding subband power ratio. In an adaptive filtering implementation, at least one of the audio signals can be adaptively filtered to allow for array self-calibration and modal-angle variability.

A voice communication device such as an earpiece that provides for ultra short range transmission of voice sound information using ultra wide band (UWB) radio communications is disclosed. According to one embodiment of the present invention, a voice communication device includes a speaker, a bone conduction sensor for sensing voice sound vibrations, and an UWB transceiver operatively connected to the speaker and the bone conduction sensor. According to another embodiment of the present invention, a voice communication device is disclosed that includes an earpiece housing, a speaker operatively connected to the earpiece housing for transducing audio, an air conduction sensor operatively connected to the housing for sensing voice sound information, a processor electrically connected to the speaker and the bone conduction sensor and disposed within the earpiece housing, and an UWB transceiver operatively connected to the processor.

A microelectromechanical-acoustic-transducer assembly has: a first die integrating a MEMS sensing structure having a membrane, which has a first surface in fluid communication with a front chamber and a second surface, opposite to the first surface, in fluid communication with a back chamber of the microelectromechanical acoustic transducer, is able to undergo deformation as a function of incident acoustic-pressure waves, and faces a rigid electrode so as to form a variable-capacitance capacitor; a second die, integrating an electronic reading circuit operatively coupled to the MEMS sensing structure and supplying an electrical output signal as a function of the capacitive variation; and a package, housing the first die and the second die and having a base substrate with external electrical contacts. The first and second dice are stacked in the package and directly connected together mechanically and electrically; the package delimits at least one of the front and back chambers.

Embodiments include methods and apparatuses for sensing acoustic waves for a conferencing application. A conferencing apparatus includes a plurality of directional microphones oriented to cover a corresponding plurality of direction vectors and disposed in a housing. An orientation sensor is configured to generate an orientation signal indicative of an orientation of the housing. A processor is operably coupled to the plurality of directional microphones and the orientation sensor. The processor is configured to automatically adjust a signal processing characteristic of one or more directional microphones of the plurality of directional microphones responsive to the orientation signal.

A semiconductor processing system including a semiconductor processing apparatus and a gas supply apparatus for supplying a process gas into the semiconductor processing apparatus includes a control section configured to control an operation of a pressure adjusting mechanism for adjusting the pressure inside a vaporizing chamber. The control section is preset to cause the pressure inside the vaporizing chamber to fall within a predetermined pressure range with reference to a pressure detection value obtained by a pressure detector. The predetermined pressure range is defined by an upper limit lower than a first limit value, at which vaporization of the liquid material starts being inhibited due to an increase in the pressure, and a lower limit higher than a second limit value, at which vaporization of the liquid material starts being unstable and the pressure inside the vaporizing chamber starts pulsating movement due to a decrease in the pressure.

In some embodiments, spatially realistic audio may be provided for a conference call. Voices from participants on the left side of a display, in a conference call, may be directed through audio on the left side of the display at the other conferencing system in the conference call (similarly for voices from the center and right side of the display). In some embodiments, two speakers may be used in the system to create synthesized stereo sound at a location specified by directional information received as side information along with the existing audio channel. The location may be determined by using beamforming with integrated microphones on a camera or speakerphone. In some embodiments, the audio signal and directional information may be sent in the form of a left audio channel and a right audio channel.

An apparatus for measuring a plurality of loudspeakers arranged at different positions includes a generator of a test signal for a loudspeaker; a microphone device configured for receiving a plurality of different sound signals in response to one or more loudspeaker signals emitted by one of the loudspeakers in response to the test signal; a controller for controlling emissions of the loudspeaker signals by the loudspeakers and for handling the different sound signals so that a set of sound signals recorded by the microphone device is associated with each loudspeaker in response to the test signal; and an evaluator for evaluating the set of sound signals for each loudspeaker to determine at least one loudspeaker characteristic for each loudspeaker and for indicating a loudspeaker state using the at least one loudspeaker characteristic. This scheme allows automatic, efficient and accurate measurement of loudspeakers arranged in a three-dimensional configuration.

Two or more microphones receive acoustic signals and generate audio signals that are processed to determine what portion of the audio signals result from (i) incoherence between the audio signals and / or (ii) audio-signal sources having propagation speeds different from the acoustic signals. The audio signals are filtered to reduce that portion of one or more of the audio signals. The present invention can be used to reduce turbulent wind-noise resulting from wind or other airjets blowing across the microphones. Time-dependent phase and amplitude differences between the microphones can be compensated for based on measurements made in parallel with routine audio system processing.

Apparatuses and methods for delivering sound to ear canal are provided. The apparatus generally includes a compressible and resilient toroidal section having an inner side and an outer side. An aperture extends from the outer side to the inner side and the outer side has a well disposed within an inner side surface about the aperture. The apparatus further includes a compressible and resilient spout integrated with the toroidal section and extending away from the inner side. The spout includes a channel disposed within an inner spout surface and the channel is coupled to the well.

Improved vocal signals are obtained in headsets and similar devices by including a microphone inside a chamber formed at least in part by the wearer's ear. This second microphone provides a reduced noise input signal. The reduced noise signal is corrected by input from another microphone, located outside the chamber. This correction can include echo cancellation, spectral shaping, frequency extension, and the like.

An apparatus comprises microphone receivers (101) which receive microphone signals from a plurality of microphones (103). A comparator (105) determines a speech similarity indication indicative of a similarity between the microphone signal and non-reverberant speech for each microphone signal. The determination is in response to a comparison of a property derived from the microphone signal to a reference property for non-reverberant speech. In some embodiments, the comparator (105) determines the similarity indication by comparing to reference properties for speech samples of a set of non-reverberant speech samples. A generator (107) generates a speech signal by combining the microphone signals in response to the similarity indications. In many embodiments, the apparatus may be distributed over a plurality of devices each containing a microphone, and the approach may determine the most suited microphone for generating the speech signal.

A detachable adjustable electronics module may be removably or permanently connected to eyewear. The module may include electronics for processing audio and / or video input and / or output signals. The module may be provided with an adjustable arm, for adjustably carrying a speaker. The module and / or the speaker may be adjusted relative to the wearer in any of the anterior-posterior direction, the inferior-superior direction and laterally. Rotation adjustments may also be accomplished. Eyewear may be provided with only a single module, on a single side, or with two modules, one on each side, such as to provide stereo audio or dual mono sound.

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.