9720results about "Transducer circuits" patented technology

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.

At least one exemplary embodiment is directed to an earpiece comprising: an ambient microphone; an ear canal microphone; an ear canal receiver; a sealing section; a logic circuit; a communication module; a memory storage unit,; and a user interaction element, where the user interaction element is configured to send a play command to the logic circuit when activated by a user where the logic circuit reads registration parameters stored on the memory storage unit and sends audio content to the ear canal receiver according to the registration parameters.

A listener can experience the sound of virtual loudspeakers over headphones with a level of realism that is difficult to distinguish from the real loudspeaker experience. Sets of personalized room impulse responses (PRIRs) are acquired for the loudspeaker sound sources over a limited number of listener head positions. The PRIRs are then used to transform an audio signal for the loudspeakers into a virtualized output for the headphones. Basing the transformation on the listener's head position, the system can adjust the transformation so that the virtual loudspeakers appear not to move as the listener moves the head.

Systems and methods for determining the operating condition of multiple microphones of an electronic device are disclosed. A system can include a plurality of microphones operative to receive signals, a microphone condition detector, and a plurality of microphone condition determination sources. The microphone condition detector can determine a condition for each of the plurality of microphones by using the received signals and accessing at least one microphone condition determination source.

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.

At least one exemplary embodiment is directed to a method of situation awareness facilitation for a vehicle driver comprising: receiving a first acoustic signal from outside a vehicle that the driver is in and converting the first acoustic signal into a first acoustic electronic signal; sending the first acoustic electronic signal to a processor; matching the first acoustic electronic signal by the processor to stored reference electronic signals; and sending an acoustic message associated with a matched reference electronic signal to a speaker in the cab of the vehicle.

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.

Audiovisual reproduction system comprising a central unit managing a sound control circuit, and a telecommunications modem connected to a distribution network controlled by a host server, through a multitask operating system created around a tools and services library, characterized in that the operating system comprises a function that adjusts the sound control circuit to couple volumes in the various areas in which the loudspeakers in the audiovisual reproduction system are used, this function being accessible through a management mode of the multitask operating system, the coupling maintaining the ratios between the various volumes in each area when the volume in one area is modified.

A glasses-type terminal including a main body configured to be worn as glasses on a user's head; a first microphone configured to detect a vibration signal through a skull portion of the user in which a voice being input propagates through the skull portion of the user; a second microphone configured to detect a voice signal in which the voice being input propagates over the air; a memory configured to store a pre-registered vibration signal and a pre-registered voice signal corresponding to a pre-registered voice; and a controller configured to switch a locked state to an unlocked state when the vibration signal detected through the first microphone is matched to the pre-registered vibration signal and the voice signal detected through the second microphone is matched to the pre-registered voice signal.

A sound source signal from a target sound source is allowed to be separated from a mixed sound which consists of sound source signals emitted from a plurality of sound sources without being affected by uneven sensitivity of microphone elements. A beamformer section 3 of a source separation device 1 performs beamforming processing for attenuating sound source signals arriving from directions symmetrical with respect to a perpendicular line to a straight line connecting two microphones 10 and 11 respectively by multiplying output signals from the microphones 10 and 11 after spectrum analysis by weighted coefficients which are complex conjugate to each other. Power computation sections 40 and 41 compute power spectrum information, and target sound spectrum extraction sections 50 and 51 extract spectrum information of a target sound source based on a difference between the power spectrum information.

An adaptive noise canceling microphone system for extracting a desired signal, in particular a desired speech signal, comprising two microphones being arranged at a predefined distance from each other; a signal forming system (SFS) being adapted to receive a first and second input signals resulting from sounds received by the two microphones wherein an acoustical signal component in the first input signal is determined, wherein an acoustical signal component in the second input signal is determined, wherein the acoustical signal component in the first input signal is enhanced to generate a speech enhanced signal, and wherein the acoustical signal component in the second input signal is suppressed to generate a speech nulled signal; an adaptive noise cancellation filtering circuit being adapted to receive the speech enhanced signal and the speech nulled signal, wherein the noise in the speech enhanced signal is cancelled using the speech nulled signal as reference, thereby generating an output filtered signal representing the desired signal.

An audio path is constructed to include a multi-bit sigma-delta converter for converting an N-bit digital input to an n-bit output representing an over-sampled, lower resolution n-bit version of the N-bit digital input; a formatter for converting the n-bit output to an m signal output (e.g., as a thermometer code, a SDM format or a PWM format); an m-by-m switching matric for receiving the m output signals and for reordering the m output signals, m class-D drivers individual ones of which are driven by one of the reordered m output signals for driving one of m speakers; and a dynamic element matching (DEM) block coupled to the switching matric for controlling the reordering of the m output signals driving the m class-D drivers for spreading the distortion due at least to driver-speaker pair mismatch to wide band noise. The DEM may operate to generate white noise, or it may generate shaped (colored) noise.

Control system for devices such as an audio reproduction system, an actuator device, an electromechanical device and a telephony device. The system includes control circuitry which receives an input signal and a signal indicative of a position of a portion of the controlled apparatus. The control circuit provides an output signal to the controlled apparatus to affect an operation of the controlled apparatus. The output signal provides control of the apparatus to compensate for one or more of: motor factor; spring factor; back electromotive force; and impedance of a coil in a driver of the controlled apparatus. The signal indicative of position is derived by one or more position indicator techniques such as an infrared LED and PIN diode combination, position dependent capacitance of one portion of the controlled apparatus with respect to another portion of the controlled apparatus, and impedance of a coil in the controlled apparatus. The control circuitry is configurable to control transconductance and/or transduction of the system being controlled. A technique is disclosed to detect and measure a cant of a voice coil transducer, the technique including measuring a capacitance between one portion of the voice coil transducer with respect to another portion of the voice coil transducer over a range of movement of the voice coil during operation.

An apparatus for detecting sound includes a plurality of microphones, a sound inspecting unit, a direction estimating unit, a background noise removing unit, and an alerting unit. The microphones are used to collect sounds around a user. The sound inspecting unit is used to calculate the feature values of a background noise within a preset time interval, and to determine if a latest collected sound satisfies a preset condition. When the preset condition is satisfied, the direction estimating unit is used to estimate the occurrence direction of the latest collected sound, and to determine if the occurrence direction is within a preset range behind the user. When the preset range is satisfied, the background noise removing unit is used to remove the background noise in the latest collected sound so as to obtain a detected sound. The alerting unit is used to inform the user of the detected sound via an alert message. A method for detecting sound is also disclosed.

Systems, methods and devices for gathering data concerning media usage by predetermined audience members. An audience measurement code in acoustic energy produced by a media receiver is detected by a portable media monitor. The monitor also detects a location code produced in a vicinity of the media receiver.

A sound collecting system includes a plurality of microphones, a distance estimation module and an adjustment module. The distance estimation module estimates a distance to a user to accordingly provide a user distance. The adjustment module adjusts a part or all of the positions of the microphones according to the user distance.

A sound measuring apparatus for measuring a sound-arrival delay time from a speaker to a microphone on the basis of a result obtained by outputting a test signal from the speaker and picking up the test signal using the microphone includes the following elements. A control unit performs control so that the test signal is expanded in a time axis and is then output from the speaker. A delay time measuring unit measures an expansion-based measured delay time on the basis of a delay time that is measured on the basis of a time difference between the test signal expanded in the time axis and output from the speaker and a signal obtained from the microphone by picking up the output expanded test signal, and obtains the sound-arrival delay time as the expansion-based measured delay time.

A method and apparatus for capturing image and sound during interactivity with a computer program is provided. The apparatus includes an image capture unit that is configured to capture one or more image frames. Also provided is a sound capture unit. The 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 zone of focus, at which to process sound to the substantial exclusion of sounds outside of the zone of focus. In this manner, sound that is captured and processed for the zone of focus is used for interactivity with the computer program.