Abstract: A method for generating an image combined with audio, and image display and audio output includes displaying an image, when a first image object within the image is selected, outputting a first audio object corresponding to the first image object and, when a second image object within the image is selected, outputting a second audio object corresponding to the second image object.

Abstract: A method and a device are provided for modifying audio signals in accordance with hearing capabilities of an individual who is listening to audio signals played by a music player. The method comprises the steps of: providing a music player operative to play audio signals, wherein the music player comprises a processor configured to modify audio signals that are about to be played, by taking into account the hearing capabilities of the individual; providing information that relates to the hearing capabilities of the individual; forwarding the information that relates to the hearing capabilities of the individual, from an electronic device to the music player; and using the music player processor to modify audio signals when the individual is listening to audio signals being played by the music player, wherein the audio signals are modified before they are played by taking into account the individual's hearing capabilities.

Abstract: An approach to notifying a person who is hard of hearing of audible events based on a configurable device. The device has microphones and associated buttons mounted on its surface. The user programs the device by depressing a selected button longer than a preconfigured time to place the device in listen mode. The user generates the desired audible event and the device records the audible event. The selected button is depressed again to instruct the device to associate the audible event with a visual alarm indicator of colored/flashing lights and/or a text projection. The device listens for the audible event and activates the visual alarm indicator when the device detects the audible event.

Abstract: According to one embodiment, an acoustic control apparatus includes an acquisition unit, a detection unit, a correction unit, and an output unit. The acquisition unit acquires a first acoustic signal. The detection unit detects an information sound. When the detection unit detects the information sound, the correction unit corrects the first acoustic signal to a second acoustic signal by convoluting the first acoustic signal with a first function. The first function represents an acoustic transfer characteristic from a virtual position to a listening position. The virtual position is located along a first direction from the listening position. The output unit outputs the second acoustic signal.

Abstract: The technology disclosed provides systems and methods for reducing the overall power consumption of an optical motion-capture system without compromising the quality of motion capture and tracking. In implementations, this is accomplished by operating the motion-detecting cameras and associated image-processing hardware in a low-power mode (e.g., at a low frame rate or in a standby or sleep mode) unless and until touch gestures of an object such as a tap, sequence of taps, or swiping motions are performed with a surface proximate to the cameras. A contact microphone or other appropriate sensor is used for detecting audio signals or other vibrations generated by contact of the object with the surface.

Abstract: The application relates to a binaural hearing assistance system comprising left and right hearing assistance devices adapted for being located at or in a left and right ear, respectively, of a user, each of the left and right hearing assistance devices comprising a) an input unit for providing one or more electric input signals based on one or more input sounds of a sound field surrounding the binaural hearing assistance system; b) a source separation unit for separating and localizing one or more sound sources Ss in said sound field relative to the input transducer unit based on said one or more electric input signals, and providing respective separated sound source signals Xs, and localization parameters LPs of said one or more sound sources (s=1, 2, . . .

Abstract: An apparatus for evaluating performance of a headphone assembly is provided with a base and a support having a proximal end extending from the base. The support includes a distal end that is adapted for supporting a headphone assembly, and the support includes a pair of opposing sides that are laterally spaced apart. The apparatus also includes a pair of plates and at least two microphones. Each plate is mounted to one of the opposing sides of the support, and includes a central portion with at least two apertures formed therein. The central portion is sized for receiving a headphone of the headphone assembly. Each microphone is disposed within one of the apertures and arranged such that an outer surface of the microphone is oriented substantially parallel to an outer surface of the plate and adapted for receiving sound emitted from the headphone.

Abstract: A method and a system of noise suppressing an audio signal comprising a combination of at least two audio system input signals each having a sound source signal portion and a background noise portion, the method and system comprising steps and means of: Extracting at least two different types of spatial sound field features from the input signals such as discriminative speech and/or background noise features, computing a first intermediate spatial noise suppression gain on the basis of the extracted spatial sound field features, computing a second intermediate stationary noise suppression gain, combining the two intermediate noise suppression gains to form a total noise suppression gain, wherein the two intermediate noise suppression gains are combined by comparing their values and dependent on their ratio or relative difference, determining the total noise suppression gain, applying the total noise suppression gain to the audio signal to generate a noise suppressed audio system output signal.

Abstract: Digital signal processing for microphone partial occlusion detection is described. In one embodiment, an electronic system for audio noise processing and for noise reduction, using a plurality of microphones, includes a first noise estimator to process a first audio signal from a first one of the microphones, and generate a first noise estimate. The electronic system also includes a second noise estimator to process the first audio signal, and a second audio signal from a second one of the microphones, in parallel with the first noise estimator, and generate a second noise estimate. A microphone partial occlusion detector determines a low frequency band separation of the first and second audio signals and a high frequency band separation of the first and second audio signals to generate a microphone partial occlusion function that indicates whether one of the microphones is partially occluded.

Abstract: A condenser microphone outputs a signal resistive to distortion caused by an excess input without a variation in signal-to-noise ratio. The microphone includes a first condenser microphone unit and a second condenser microphone unit that generate output signals having phases opposite to each other; and a twin variable resistor connected to an output terminal of each of the first and second condenser microphone units. In the microphone, the output signal of the first unit is combined with the output signal of the second unit in a first variable resistor included in the twin variable resistor and the composite signal is applied to an input of the first unit, and the output signal of the first unit is combined with the output signal of the second unit in a second variable resistor included in the twin variable resistor and the composite signal is applied to an input of the second unit.

Abstract: A microphone 1 includes a circuit board 22 equipped with electronic circuits, a microphone unit 26, and a case 21 accommodating the circuit board 22 and the microphone unit 26. The case 21 is a resilient hollow cylinder. The circuit board 22 has a substantially rectangular shape and has shorter sides having a length substantially equal to the inner diameter of the case 21. The circuit board 22 includes contact adapters 222 in contact with the inner wall of the case 21, along the longer sides of the circuit board 22.

Abstract: An apparatus and a method for sound recording are provided. The apparatus stores an equalizer and includes a housing, a microphone, and a processor. The microphone includes a plurality of sensors and receives a sound signal through an acoustical resonator in the housing. The microphone generates a plurality of electronic signals in response to the sound signal. The equalizer generates a plurality of equalized signals according to the electronic signals. The processor generates an output signal according to the equalized signals. The equalizer compensates the gain margin and the phase margin caused by the sound signal passing through the acoustical resonator in order to prevent the output signal from being affected by resonance of the sound signal in the acoustical resonator.

Abstract: The invention is concerned with headphones having left and right supporting members (9, 19) for a sound transducer (11, 14, 20). In order to allow for an ameliorated out-of-head localization of acoustic or sound events without noteworthy loss of treble and midrange information in a hearing protecting manner the sound transducer (11, 14, 20) occupies a position tilted about a horizontal axis in the viewing direction of a headphones user and on a vertical axis intersecting the horizontal axis.

Abstract: An apparatus for determining a position estimate for an audio source comprises two microphones (M1, M2) and an acoustic element (203) providing an acoustic effect to sound from sound source positions to the first microphone (M1). The acoustic effect is asymmetric with respect to an axis (201) between the microphones (M1, M2). A position circuit (305) estimates two possible positions on different sides of the axis for the sound source in response to time of arrivals at the microphones (M1, M2). An estimator (307) determines an acoustic effect measure being indicative of a degree to which an acoustic effect of the first microphone signal matches an expected characteristic of the acoustic effect for sound sources on one side of the axis (201). Another circuit (309) determines the position estimate by selecting between the two possible positions in response to the acoustic effect measure.

Abstract: A system, method and related devices for monitoring and improving the training and social eye contact and communication skills of developmentally challenged individuals such as autistic individuals with special needs in improving their interpersonal communicating and other skills. A WatchMe component of the system and method monitors and obtains qualitative and quantitative information about the eye contact habits of a subject being trained or interviewed. It provides stimuli that promotes and encourages improvements in the eye contact habits of the subject.

Abstract: There is provided a microphone comprising a microphone capsule and a microphone capsule suspension. The microphone capsule suspension has a first and a second housing portion and a decoupling unit. In that case the microphone capsule is mounted to the decoupling unit for solid-borne sound decoupling. The decoupling unit is in turn fixed to the first and/or second housing portion. The decoupling unit has a shape-changing portion which under loading changes its shape. The decoupling unit has at least two arms which each have a first and a second end. The second ends of the arms respectively have a fixing portion for receiving a microphone capsule. Each arm has a shape-changing portion which changes its shape under a first loading and an elastic portion which permits elastic deformation in respect of length as from a second loading.

Abstract: 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.

Abstract: A method and a device for eliminating or minimizing the sensitivity changes in a microphone due to temperature changes. The temperature-induced changes in the sensitivity can be caused by the changes in the sound-to-electrical signal transducer, in the microphone membrane, in the ASIC or other reasons. One or more temperature dependent components in the microphone or in a microphone module are used to offset the temperature-induced changes in the sensitivity. Sensitivity of a microphone is defined as the output voltage for a specific acoustic stimulus and load condition.

Abstract: Methods and apparatus to detect spillover in an audience monitoring system are disclosed. An example method includes sampling a first audio signal from a first source received by a first microphone; sampling a second audio signal received by a second microphone from the first source, wherein the second microphone is separated from the first microphone by a first distance; calculating an origin direction of the first source based on a time difference between receipt of the first and second audio signals; when the origin direction is within a threshold angle, retaining the media associated with the first source; and when the origin direction exceeds the threshold angle, ignoring the media associated with the first source.

Abstract: Described herein are techniques for artificially enhancing spaciousness in a hearing aid to improve the music listening experience. Such spatial enhancement is produced by doing signal processing in the hearing aid that mimics the acoustic effects of well-designed concert halls. The same techniques can also be applied to improving the experience of listening to recorded music reproduced and amplified over a speaker system, or to music streamed to the direct-audio input of a hearing aid.

Abstract: A condenser stereomicrophone includes a mid condenser microphone unit that is connected to left and right side condenser microphone units such that the output from the mid condenser microphone unit is applied to the left and right side condenser microphone units. The left and right side condenser microphone units are connected to left and right-channel connector terminals, respectively; and the left and right-channel connector terminals are connected to the mid condenser microphone unit via corresponding power-supply resistors.

Abstract: A method and system for obtaining an audio signal. In one embodiment, the method comprises receiving a first sound signal at a first microphone arranged at a first height vertically above a substantially flat surface; receiving a second sound signal at a second microphone arranged at a second height vertically above the substantially flat surface; processing a signal provided by the first microphone using a low pass filter; processing a signal provided by the second microphone using a high pass filter; adding the signals processed by the low pass filter and the high pass filter to form a sum signal; and outputting the sum signal as an audio signal.

Abstract: A stereo microphone includes a nondirectional microphone unit and two bidirectional microphone units, the bidirectional microphone units having directional axes disposed on a single plane at predetermined angles to the center axis of the directionality of the stereo microphone, an acoustic terminal of the nondirectional microphone unit being disposed in the vicinity of acoustic terminals of the bidirectional microphone units.

Abstract: The present application provides a system, method and non-transitory computer readable medium that provides a means of using a vehicle horn as an inexpensive user input interface to serve the function of a simple push button switch for an in-vehicle device or system. The use of the vehicle horn as a user input interface to an in-vehicle device or system is novel. The description of example embodiments illustrates application details that take advantage of the properties of the vehicle horn sound to allow efficient processing that can be implemented on a low cost processor.

Abstract: Two ear cups are coupled by a flexible member. Each ear cup includes a speaker mounted on an internal portion thereof and is configured to form an air pressure seal around an ear of a user when the apparatus is placed on the user. The flexible member couples the two ear cups and ijs configured such that, when the apparatus is placed on the user, each ear cup completely covers an associated ear of the user and forms an air pressure seal around the associated ear. A microphone is electrically coupled to the speaker in each ear cup for transmitting ambient sound to the user when the apparatus is placed on the user. An indicator light may be provided which is configured to turn on when the microphone is active. The microphone and/or the indicator light may be mounted on the flexible member or on one of the ear cups.

Abstract: An apparatus comprises at least one processor, and at least one memory including computer program code, the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus to: provide an audio capture event wherein one or more microphone configurations are configured to provide one or more audio signals based on at least one acoustic signal from at least one acoustic source, at least one of the one or more microphone configurations being defined by a first position of a first microphone configuration on a first portion and a second position of a second microphone configuration on a second portion, the second portion being movable relative to the first portion.

Abstract: A system enhances spatialization in which spatial information about sound sources at an originating location is represented in an audio signal. The system applies a phase difference analysis to the signals received from an array of spaced apart input devices or microphones to derive spatial or directional information about the relative directions of one or more satellite input devices or microphones. The signals from the satellite input devices or microphones are mixed by a function of their respective directions to generate a multichannel output signal. When processed by a remote or local system, the output signal provides a representation of the relative directions of the sound sources at the originating location at a receiving location.

Abstract: The invention relates to a portable electronic device, comprising: at least two directional microphones for stereo sound pickup, each one of the two directional microphones defining a direct sound direction and an opposite sound direction towards which the directional microphones are directed; and a housing comprising for each of the directional microphones a first hole and a second hole, the first hole being located at a different side of the portable electronic device than the second hole.

Abstract: A stereo microphone has four condenser microphone units having respective directional axes in the same horizontal plane. The four units each have unidirectivity and a quadrangular shape viewed from the direction of the directional axis. The units are disposed by rotating the directional axes of adjacent units by 90°. A pair of two units diagonally positioned and having the directional axes directed at 180° to each other collaborate with each other to form a pair of bidirectional microphone units.

Abstract: A microphone array, comprising N microphones, wherein N is greater than or equal to 3 is provided. The microphones are substantially equiangularly arranged over a circular arc subtending an angle ?, wherein ? is less than or equal to 2?, with the directional axes of the N microphones facing substantially radially outwards. Each of the N microphones have a substantially common directivity function ?(?) defining the directional response of the microphone, wherein ?=0 is the directional axis, and the directivity function ?(?) is arranged such that a sound source in acoustical free field is effectively captured by no more than two consecutive microphones in the array. By arranging the directivity function in this manner crosstalk between non-adjacent microphones can be minimized, which has been shown to improve auditory localization performance.

Abstract: A personal audio device, such as a wireless telephone, includes noise canceling circuit that adaptively generates an anti-noise signal from a reference microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone may also be provided proximate the speaker to estimate an electro-acoustical path from the noise canceling circuit through the transducer. A processing circuit uses the reference and/or error microphone, optionally along with a microphone provided for capturing near-end speech, to determine whether one of the reference or error microphones is obstructed by comparing their received signal content and takes action to avoid generation of erroneous anti-noise.

Abstract: An apparatus, method, and medium of removing crosstalk are provided. The apparatus for removing crosstalk in each of audio signals of a plurality of channels, includes: a position recognition unit recognizing the position of a listener; a calculation unit calculating a crosstalk removal function with respect to the recognized position; and a filtering unit removing the crosstalk by using the calculated result.

Abstract: An audio processing system may modify an input surround sound signal to generate a spatially equilibrated output surround sound signal that is perceived by a user as spatially constant for different sound pressures of the surround sound signal. The audio processing system may determine based on a psychoacoustic model of human hearing, a loudness and a localization for a combined sound signal. The loudness and the localization may be determined by the system for a virtual user located between the front and the rear loudspeakers that has a predetermined head position in which one ear of the virtual user is directed towards one of front or rear loudspeakers and the other ear of the virtual user being directed towards the other of the front or rear loudspeakers. The audio processing system may adapt the front and/or rear audio signal channels based on the determined loudness and localization.

Abstract: An audio playback system includes a handheld device and a digital stereo set. By the handheld device, a control message is converted into a modulation signal. The modulation signal is transmitted to a speaker, thereby controlling the speaker to generate a modulated sound wave signal. After the modulated sound wave signal is received by a built-in microphone of the digital stereo set, the modulated sound wave signal is restored to the control message. After the control message is received by the digital stereo set, the digital stereo set performs a corresponding control action.

Abstract: A method for generating engine acoustic emissions including the steps of providing a loudspeaker configuration (100) based upon the configuration of said engine, and feeding the said loudspeaker configuration (100) with signals for providing desired engine acoustic emission. Also a device, a computer program and a computer program product for performing the method.

Abstract: A method and apparatus for localizing a sound image of an input signal to a spatial position are provided. The method of localizing a sound image to a spatial position includes: extracting from a head related impulse response (HRIR) measured with respect to changes in the position of a sound source, first information indicating a reflection sound wave reflected by the body of a listener; extracting from the HRIR second information indicating the difference between sound pressures generated in two ears, respectively, when a direct sound wave generated from the position of the sound source arrives at the two ears, respectively, of the listener; extracting third information indicating the difference between times taken by the direct sound wave to arrive at the two ears, respectively, from the HRIR; and localizing a sound image of an input signal to a spatial position by using the extracted information.

Abstract: A stereo microphone includes two unidirective mid units and a bidirective side unit, the side unit is a ribbon microphone unit including a ribbon diaphragm, the two mid units are disposed at two respective surfaces of the ribbon diaphragm of the side unit, and the mid units each have a sound collecting axis along a longitudinal direction of the ribbon diaphragm in the side unit.

Abstract: A method for switching recording modes using an electronic device requires a three-axis acceleration senor and at two microphones. The three-axis accelerometer is activated to detect three-axis acceleration values of the electronic device. A handheld direction of the electronic device is determined according to the detected three-axis acceleration values. The method set a recording software of the electronic device to select one of the recording modes to execute a recording function according to the determined handheld direction. The method further selects the first microphone and/or the second microphone to receive external audio signals according to the selected recording mode.

Abstract: A sound zooming technique synchronized with a moving picture zooming function is disclosed. A signal to control ambient noise and a signal to control a long distance sound are extracted from an audio signal input through a plurality of microphones. A front directivity-emphasized signal is masked using the two signals and a weighted value to control volume.

Abstract: An audio communication system includes a headset with a microphone having two electrical outputs and a first electro-acoustic driver. The system includes first and second differential amplifiers, and a first electrical conductor for electrically connecting an output of the first differential amplifier with a first audio source device. A second electrical conductor is provided for electrically connecting an input of the driver with the first audio source device. A third electrical conductor is provided for electrically connecting an output of the second differential amplifier with a second audio source device. A fourth electrical conductor is provided for electrically connecting an input of the driver with the second audio source device. One electrical output from the microphone is electrically connected to a positive input of each of the differential amplifiers. The other electrical output from the microphone is electrically connected to a negative input of each of the differential amplifiers.

Abstract: A disclosed method selects a plurality of fewer than all of the channels of a multichannel signal, based on information relating to the direction of arrival of at least one frequency component of the multichannel signal.

Abstract: A method for enhancing speech includes extracting a center channel of an audio signal, flattening the spectrum of the center channel, and mixing the flattened speech channel with the audio signal, thereby enhancing any speech in the audio signal. Also disclosed are a method for extracting a center channel of sound from an audio signal with multiple channels, a method for flattening the spectrum of an audio signal, and a method for detecting speech in an audio signal. Also disclosed is a speech enhancer that includes a center-channel extract, a spectral flattener, a speech-confidence generator, and a mixer for mixing the flattened speech channel with original audio signal proportionate to the confidence of having detected speech, thereby enhancing any speech in the audio signal.

Abstract: A compact portable stereo microphone apparatus can acquire three dimensional immersive sounds which can then be recorded and played back in a standard audio recording device. The apparatus includes two elastomer fabricated simplified human ear elements (ear models) fitted at the ends of a cylindrical cavity of a microphone enclosure. The ear models may be protected against physical damage by end grills. A circuit board, microphone cable interface, power indicator, power switch and batteries may be located in the lower shell of the enclosure. A handle may serve as the hand grip for the microphone as well as a battery cover.

Abstract: A home use sound reproduction system for Hi-fi, with digital signal transfer from a playback unit via a control unit to one or more active loudspeakers, each including or arranged beside an amplifier unit. The control unit is arranged to control sound parameters and send both a digital sound information signal and a power signal for powering the amplifier units, and to superimpose the single ended or differential digital signal together with the power signal on at least one common lead in a cable.

Abstract: A remote controlled robot system that includes a robot and a remote control station. The robot includes a binaural microphone system that is coupled to a speaker system of the remote control station. The binaural microphone system may include a pair of microphones located at opposite sides of a robot head. the location of the microphones roughly coincides with the location of ears on a human body. Such microphone location creates a mobile robot that more effectively simulates the tele-presence of an operator of the system. The robot may include two different microphone systems and the ability to switch between systems. For example, the robot may also include a zoom camera system and a directional microphone. The directional microphone may be utilized to capture sound from a direction that corresponds to an object zoomed upon by the camera system.

Abstract: A low-noise pre-amplifier with an active load element is integrated into a microphone. The microphone has an acoustic sensor coupled to the intrinsic pre-amplifier. A controllable current source is coupled to the intrinsic pre-amplifier and supplies a pre-amplifier bias current. A current source controller is coupled to the current source and controls the amplitude of the pre-amplifier bias current to maintain the intrinsic pre-amplifier at a bias point at which the intrinsic pre-amplifier amplifies microphone signals produced by the acoustic sensor. The intrinsic pre-amplifier may be actively regulated at the pre-determined bias point using negative feedback. Alternatively, the intrinsic pre-amplifier may be set to the pre-determined bias point by sweeping the pre-amplifier bias current for the intrinsic pre-amplifier over a range of currents.

Abstract: A recorder (10) for recording a scene (12) includes an apparatus frame (218), an optical assembly (220), an image system (222), a position assembly (243), an audio system (224), and a compensation system (248). The image system (222) captures an image (252) of the scene (12). The position assembly (243) can be an autofocusing assembly (244) that focuses the optical assembly (220) on a subject (16) of the scene (12). The position assembly (243) generates position information relating to the position of the subject (16) relative to the recorder (10). The audio system (224) captures a captured sound from the scene (12). The compensation system (248) evaluates the position information and the captured sound from the scene (12) and provides an adjusted sound track in view of the position information.

Abstract: Spherical microphone arrays capture a three-dimensional sound field for generating an Ambisonics representation, where the pressure distribution on the surface of the sphere is sampled by the capsules of the array. The impact of the microphones on the captured sound field is removed using the inverse microphone transfer function. The equalisation of the transfer function of the microphone array is a big problem because the reciprocal of the transfer function causes high gains for small values in the transfer function and these small values are affected by transducer noise. The invention minimises that noise by using a Wiener filter processing in the frequency domain, which processing is automatically controlled per wave number by the signal-to-noise ratio of the microphone array.

Abstract: An augmented reality (AR) audio system for augmenting environment or ambient sound with sounds from a virtual speaker or sound source positioned at a location in the space surrounding an AR participant. The sound from the virtual speaker may be triggered by an action of the listener and/or by the location or relative orientation of the listener. The AR audio system includes stereo earphones receiving an augmented audio track from a control unit, and binaural microphones are provided to capture ambient sounds. The control unit operates to process trigger signals and retrieve one or more augmentation sounds. The control unit uses an AR audio mixer to combine the ambient sound from the microphones with the augmentation sounds to generate left and right ear augmented audio or binaural audio, which may be modified for acoustic effects of the environment including virtual objects in the environment or virtual characteristics of real objects.

Abstract: A condenser stereomicrophone includes a mid condenser microphone unit that is connected to left and right side condenser microphone units such that the output from the mid condenser microphone unit is applied to the left and right side condenser microphone units. The left and right side condenser microphone units are connected to left and right-channel connector terminals, respectively; and the left and right-channel connector terminals are connected to the mid condenser microphone unit via corresponding power-supply resistors.