Abstract: A method and a device for presenting multimedia information are disclosed. The method includes: acquiring representation information, the representation information including electromagnetic-field spectral information for representing an object, the electromagnetic-field spectral information being observable for a naked eye and/or acquirable for a device; establishing a four-dimensional time-space model for characterizing the representation information according to the acquired representation information, the four-dimensional time-space model having an attribute for characterizing in a digital form variation of the representation information over time; and presenting the representation information characterized by the four-dimensional time-space model. In the solution, the four-dimensional time-space model has an attribute for characterizing in a digital form variation of the representation information over time.

Abstract: A device including an array of bidirectional microphones optimizes the echo rejection of the bidirectional microphones. The microphone array receives audio from an audio source and generates audio signals from each of the bidirectional microphones. The device forms audio beams from combinations of the audio signals generated from the microphone array. Each audio beam captures audio from either its positive polarity zone or its negative polarity zone. The device determines a direction of the audio source and selects a perpendicular audio beam pair based on the direction of the audio source. The perpendicular audio beam pair includes a primary audio beam aimed toward the direction of the audio source and a secondary beam perpendicular to the primary audio beam. The device generates an output signal by combining the primary audio beam with the secondary audio beam based on polarity zone the audio is captured for each audio beam.

Abstract: A hearing assistance system having a first hearing device worn at a first of a user's ears, a second hearing device worn at a second of the user's ears, and an external audio source device for transmitting an external audio data stream, each hearing device having an interface for wireless data exchange with the external device, including receipt of the external audio data stream, the hearing devices being adapted to establish a binaural link between the first and second hearing devices, the hearing assistance system being adapted to estimate the quality of at least one of the links from the external device to the hearing devices. The hearing assistance system is adapted to dynamically adjust the coding of the audio stream transmitted by the external device according to the estimated quality of the at least one of the links from the external device to the hearing devices.

Abstract: The present invention comprises a microphone simulator or emulator, for use in conjunction with a “personal electronic device” or “PED.” The microphone simulator or emulator is used in conjunction with a PED that includes both a built-in microphone, along with an interface by which an external microphone can be utilized. Often, a PED is equipped with no specific user-level interface, by which the built-in microphone of the PED can be disabled. However, the microphone simulator or emulator, when connected to the external microphone interface of a PED, causes the PED to recognize the microphone simulator or emulator as an external microphone. Because of this, the PED disables its built-in microphone.

Abstract: A system including a transducer and an amplifier as well as a circuit which always has a high impedance at low voltages. In addition, at high voltages, the circuit has a high impedance at high frequencies but a low impedance at low frequencies. In biased transducers, this circuit may be used between the charge pump and the transducer. In general, the circuit may be provided in a signal path between the transducer and the amplifier. The circuit has as an advantage that at startup, low frequency signals at high intensities may overload the amplifier, whereas at operation, higher frequency signals are desired fed to the amplifier at the same intensity. This is facilitated by the circuit.

Abstract: A turkey sensor that is used to scout an area without the hunter being present. The device locates wildlife by recording their vocals primarily turkey and duck roosts in the first 30 minutes of daylight in the early morning.

Abstract: A platform for a military radio with a vehicle adapter amplifier has been developed. The apparatus includes a base for supporting an AN/PRC-117G Man Pack radio in combination with a SINCGARS RT-1523 radio, a for electrically connecting with the AN/PRC-117G Man Pack radio and the SINCGARS RT-1523 radio, and a vehicular power supply system.

Abstract: A computing device is provided, comprising a processor configured to receive a set of measurements of a vector x of acoustic data, including noise, interference, and a signal of interest. The processor may express x in a frequency domain discretized in a plurality of intervals. For each interval, the processor may generate an estimate ?x of a covariance matrix of x. For each ?x, the processor may use acoustic imaging to obtain an estimate ? of a spatial source distribution. For each ?, the processor may remove the signal of interest to produce an estimate ? of a noise and interference spatial source distribution. For each ?, the processor may generate an estimate ?n of a noise and interference covariance matrix. The processor may generate a beamformer configured to remove noise and interference from the acoustic data, wherein the noise and interference at each frequency are identified using ?n.

Abstract: A pre-amplifier circuit including a microphone pre-amplifier stage with a direct injection (DI) unit that provides a high impedance source signal and a low impedance XLR connection operable without a line or mic switch.

Abstract: A computer-implemented system and method for identifying and locating an emitted acoustic signal from a source entity is disclosed. The system and method is capable of distinguishing between source entities of the same type. The system and method may further allow a user of the system to observe the movement of a source entity in real-time from a remote location as well as access stored data representative of the movement of the source entity from a remote location. The system and method is capable of identifying the identity of a source entity by using an average sound source comparing a received emitted acoustic signal from the source entity to a database of sounds. The system and method is capable calculating the source entity's distance and direction from an origin input sensor for each occurrence of emitted acoustic signals received from the source entity.

Abstract: An electronic device includes a plurality of microphones, each pair of microphones in the plurality of microphones being a respective distance from one another. The electronic device also includes a beamformer functional block that receives audio signals from each of the microphones. The beamformer functional block detects a dominant microphone from among the plurality of microphones based on the audio signals from each of the microphones and the distances between the microphones, the dominant microphone being in a closest direction to a source of desired audio. The beamformer functional block also detects interfering audio signals based on phase coherence between audio signals from the dominant microphone and audio signals from other microphones in the plurality of microphones. The beamformer functional block generates a beamformed audio output signal based on the audio signals and the interfering audio signals from each of the microphones.

Abstract: Some embodiments of the invention include a throwable microphone device. The throwable microphone device may comprise a housing. The throwable microphone device may include a microphone, a communication unit, a motion sensor, an orientation sensor, and a processor disposed within the housing. In some embodiments, the microphone may receive sound waves and generate a corresponding electrical audio signal. The communication unit may wirelessly transmit at least a portion of the electrical audio signals. The motion sensor may detect changes in acceleration of the throwable microphone device. The orientation sensor may detect changes in orientation of the throwable microphone device. The processor may be electrically coupled with the microphone, the communication unit, the motion sensor, and the orientation sensor. The processor may mute the throwable microphone device in response to data from the motion sensor and may also unmute the throwable microphone device in response to data from the orientation sensor.

Abstract: Some aspects of the disclosure relate to a system and methods for monitoring an audio signal that may comprise one or more device audio components, e.g., from a device presenting content and an ambient audio component. The system can determine the ambient audio component of the audio signal, and can determine whether the ambient audio component has satisfied an ambient audio component threshold. If the ambient audio component threshold has been satisfied, the system can cause the device to display a visual representation of the device audio component, such as presenting text corresponding to speech of the audio component.

Abstract: A method and apparatus are provided that are configured to initiate a sound capture event within an apparatus and to receive at least two audio signals during the sound capture event. The at least two audio signals are provided by at least two microphones associated with the apparatus. The method and apparatus are also configured to determine at least one microphone operational parameter based on the received at least two audio signals; and to control the sound capture event such that at least one of the at least two audio signals is processed based on the at least one microphone operational parameter.

Abstract: Systems, methods, and apparatus for facilitating multi-sensor signal optimization for speech communication are presented herein. A sensor component including acoustic sensors can be configured to detect sound and generate, based on the sound, first sound information associated with a first sensor of the acoustic sensors and second sound information associated with a second sensor of the acoustic sensors. Further, an audio processing component can be configured to generate filtered sound information based on the first sound information, the second sound information, and a spatial filter associated with the acoustic sensors; determine noise levels for the first sound information, the second sound information, and the filtered sound information; and generate output sound information based on a selection of one of the noise levels or a weighted combination of the noise levels.

Abstract: A surgical helmet assembly includes a surgical helmet, headphones mounted on the surgical helmet, and a noise cancellation system configured to supply a noise cancelling audio input to the headphones to mitigate unwanted ambient noise. A gain of the noise cancelling audio input is configurable to permit necessary feedback noise from surgical tools or the like to be heard by the wearer.

Abstract: A computing device may include a multi-functional port in a base housing of the device. The multi-functional port may include an audio input device that can receive and process external audio input signals through an opening in the base housing. The multi-functional port may include a light source that can output light through the opening in the base housing. The multi-functional port may include a microphone boot. The microphone boot may guide external audio input signals into to the audio input device for processing. The microphone boot may also guide light, emitted by the light source, out of the base housing.

Abstract: A method, apparatus and a computer storage medium for selecting a microphone are disclosed. The method includes: employing ultrasonic measurement to determine a microphone, which is closest to a primary sound source, of a matrix of a plurality of microphones arranged in a device for recording sound; and taking the microphone which is closest to the primary sound source as the current primary microphone and taking other microphones of the matrix of the plurality of microphones as secondary microphones, wherein the primary microphone is used to collect the primary sound source, and the secondary microphones are used to collect ambient noise.

Abstract: Various embodiments provide for an acoustically configurable microphone device that can adjust various parameters affecting sensitivity, acoustic overload point, signal to noise ratio, sample rate, and other parametric choices that affect the performance and function of the acoustic sensor based on commands received via acoustic communications. An acoustic analysis engine can analyze acoustic communications received to determine whether or not the acoustic communication contains a command associated with configuring the microphone. If the acoustic communication is determined to contain a configuration command, a controller on the microphone can implement the configuration command. The acoustic communications can be at ultrasonic frequencies in some embodiments so that the acoustic communications are outside the range of human hearing.

Abstract: An audio device may have an output speaker that produces audio within the environment of a user and one or more input microphones that capture speech and other sounds from the user environment. The audio device may use acoustic echo cancellation (AEC) to suppress echoed components of the speaker output that may be present in audio captured by the input microphones. The AEC may be implemented using an adaptive filter that estimates echoing based on an output reference signal. The output reference signal may be generated by a reference microphone placed near the speaker of the audio device.

Abstract: An electronic device is provided. The electronic includes a camera module and a processor configured to capture at least one image of an object using the camera module, obtain a sound using a microphone operably connected to the processor when the at least one image is captured, determine whether the sound is related to the object, when the sound is determined to be unrelated to the object, change at least one attribute of the sound and store the changed at least one attribute of the sound.

Abstract: A microphone device is provided, including first and second chambers, first and second acoustic sensors, and a sound transmission device. The first and second chambers include the first and second acoustic ports, respectively. The first and second acoustic sensors are arranged in the first chamber and the second chamber, respectively. The sound transmission device coupled to the first and second chambers includes third and fourth acoustic ports, a first acoustic tube, and a second acoustic tube. The first acoustic tube communicates with the first acoustic port and the third acoustic port. The second acoustic tube communicates with the second acoustic port and the fourth acoustic port. The sensitivity difference between the first acoustic sensor and the second acoustic sensor is determined based on the length difference or the cross-sectional area difference between the first acoustic tube and the second acoustic tube.

Abstract: A method (500) for constructing a three-dimensional (3D) wave field representation of a 3D wave field using a two-dimensional (2D) sensor array (110), said method comprising: acquiring 3D wave field signals using a 2D array (110) of sensors (340, 350), said 2D array (110) of sensors (340, 350) comprising omnidirectional sensors (340) and first order sensors (350) arranged in a 2D plane; digitizing said acquired 3D wave field signals; computing even coefficients of spherical harmonics dependent upon said digitized 3D wave field signals acquired by said omnidirectional sensors (340); computing odd coefficients of said spherical harmonics dependent upon said digitized 3D wave field signals acquired by said first order sensors (350); and constructing a 3D wave field representation dependent upon said computed even and odd coefficients for said acquired 3D wave field signals.

Abstract: A processor system includes first and second regulators for regulating an adjusted supply voltage. In one embodiment, the regulator system comprises a digital low-dropout (DLDO) control system comprising first and second regulators that generate a plurality of control signals to regulate an adjusted power supply voltage and that generate a charge when a droop level falls below a droop threshold value. The first regulator implements a first control loop and the second regulator implements a second and much faster acting control loop. A supply adjustment block with the two regulators and control loops are provided for each processor core allowing different cores to have different regulated supply levels all based on one common supply.

Abstract: A two-scale array for detecting wind noise signals and acoustic signals includes a plurality of subarrays each including a plurality of microphones. The subarrays are spaced apart from one another such that the subarrays are configured to detect acoustic signals, and the plurality of microphones in each subarray are located close enough to one another such that wind noise signals are substantially correlated between the microphones in each subarray.

Abstract: A microphone device that can be thrown to a member of an audience or to another user so that the microphone device recognizes the state of the motion of the microphone device and switches from a first state of motion to another state of motion when the state of motion is changed during the throw and switches the state of the microphone device from a mute state to unmute state according to the state of the motion.

Abstract: Methods and apparatus include headphones that determine a size of a head of a listener and a shape of an ear of the listener. The size of the head and the shape of the ear determine head-related transfer functions (HRTFs) that are customized to the listener. The headphones include head tracking, and sound is adjusted so that a location of a source of the sound continues to originate from a fixed sound localization point (SLP) in empty space that is at least one meter away from the head of the listener while the head orientations of the listener change with respect to the fixed location.

Abstract: The described examples provide for a sound recognition and notification process and apparatus that provides a notification to a hearing impaired person via a portable device in response to the recognition of a selected sound. A processor of the sound notification apparatus is configured to compare the detected sound data to a model for recognition. If the sound data matches the model, features are extracted from the detected sound data and used by the processor to generate reference data, e.g. an updated sound model. The reference is retained in a memory of the apparatus to facilitate subsequent sound recognition without having to rely upon an external device for analysis. Recognition of a sound, based on a substantial match to stored reference data, may trigger transmission of a notification message indicating an occurrence of the corresponding sound.

Abstract: A data-processing method for determining at least one spatial coordinate of a sound source emitting a sound signal, in a three-dimensional space, includes the following steps: obtaining at least one first signal and one second signal from the sound signal, collected according to separate directivities by a first sensor and a second sensor; deducing from the first and second signals an expression of at least one first spatial coordinate of the sound source, the expression comprising an uncertainty; determining additional information relating to the first spatial coordinate of the sound source, from a comparison between the respective features of the signals collected by the first and second sensors; and determining the first spatial coordinate of the sound source on the basis of the expression and the additional information.

Abstract: The present invention relates to a sound effect control method and apparatus, where the method includes: when a hands-free call is performed for a mobile terminal, detecting whether a hands-free call channel of the mobile terminal is shielded; and adjusting a configuration of the hands-free call channel of the mobile terminal and/or outputting an alarm signal to inform a user that the hands-free call channel is shielded, when it is detected that the hands-free call channel of the mobile terminal is shielded. According to the method, when a hands-free call is performed for a mobile terminal, whether a hands-free call channel of the mobile terminal is shielded is detected; when it is shielded, a configuration of the hands-free call channel of the mobile terminal is adjusted and/or an alarm signal is output, which can effectively improve quality of the hands-free call of the mobile terminal.

Abstract: Electronic devices in a computer system execute to provide a sound localization point in empty space of a voice during an electronic call between a first person and a second person. The sound localization point occurs at least one meter away from a head of the person wearing a wearable electronic device at an augmented reality image.

Abstract: A method executed in a computer system that provides voices of a first person and a second person in a telephone call to each other in a computer-generated space. The voices localize at a sound localization point in empty space that is at least one meter away from the person.

Abstract: A microphone porting and venting assembly (100) is formed of a remote support substrate (118) providing a (130) acoustically resistive element with dedicated venting cavities (132) along with an external baffle (220) providing acoustic channels (212, 214, 216) which further provide water drainage and external sound sampling points (222, 224, 226). The microphone porting and venting assembly (100) is well suited waterproof, noise cancelling microphone systems.

Abstract: A method executed by electronic devices that provide voices of two people in a Voice over Internet Protocol (VoIP) call as binaural sound. The electronic devices process the voices with head-related transfer functions (HRTFs) so the voices externally localize to the two people at least one meter away in empty space where no physical object exists.

Abstract: A treadmill having a running deck comprising a motor arranged to drive movement of a tread belt, a processor, memory in electronic communication with the processor, and instructions stored in the memory. The instructions are executable by the processor to determine an anti-phase waveform based on waveform attributes of a noise emitted from the treadmill and to cause a sound of the anti-phase waveform to be emitted into a surrounding environment.

Abstract: A sound processing apparatus includes a sound determination portion operable to determine whether an input sound includes a first sound emitted from a particular source based on location information of the source, a sound separation portion operable to separate the input sound into the first sound and a second sound emitted from a source different from the particular source if the sound determination portion determines that the input sound includes the first sound, and a sound mixing portion operable to mix the first sound and the second sound separated by the sound separation portion at a prescribed volume ratio.

Abstract: A sound processing apparatus includes a first calculator that calculates first power based on a first signal received by a first microphone that is among the first microphone and a second microphone; a second calculator that calculates second power based on a second signal received by the second microphone; a gain calculator that calculates a gain on the basis of the ratio of the first power to the second power; and a multiplier that processes the second signal using the gain calculated by the gain calculator.

Abstract: A high-performance method evaluates a useful signal of an audio device, and in particular of an audio apparatus, for example for reducing interference. Accordingly, in the method at least two microphone signals are each obtained from a sound signal and a reference signal is obtained from the microphone signals, a portion of the microphone signals from a predetermined direction being blocked. The microphone signals are filtered by a filter such that an evaluation signal is obtained. To that end, a coherence value is determined from portions of the reference signal and a power density value is determined from the coherence value. The filter is parameterized on the basis of the power density value.

Abstract: An apparatus for testing a directional hearing instrument includes: a first microphone for coupling with an output of the hearing instrument, wherein the first microphone is configured to receive an audio output signal from the hearing instrument; a first speaker for transmission of a first signal having a first frequency component at a first frequency; a second speaker for transmission of a second signal having a second frequency component at a second frequency; and a processing unit configured to determine one or more hearing instrument parameters based on cross spectrum analysis of the first signal and the audio output signal.

Abstract: Embodiments disclosed herein enable detection and improvement of the quality of the audio signal using a mobile device by determining the loss in the audio signal and enhancing audio by streaming the remainder portion of audio. Embodiments disclosed herein enable an improvement in the sound quality rendered by rendering devices by emitting an test audio signal from the source device, measuring the test audio signal using microphones, detecting variation in the frequency response, loudness and timing characteristics using impulse responses and correcting for them. Embodiments disclosed herein also compensate for the noise in the acoustic space by determining the reverberation and ambient noise levels and their frequency characteristics and changing the digital filters and volumes of the source signal to compensate for the varying noise levels.

Abstract: Embodiments include a method and an apparatus for the localization of at least one source of an acoustic signal including: temporally sampling the acoustic signal with a plurality of microphones to obtain a (D+1)-dimensional space-time matrix representation of the acoustic signal, wherein D is the number of spatial dimensions, applying a (D+1)-dimensional Fourier transform to the matrix representation, determining a first peak in a spectrum obtained based on the application of the Fourier transform, and calculating the direction of arrival of the acoustic signal at at least one of the plurality of microphones based on the determined first peak.

Abstract: Accordingly, the present disclosure is directed to an audio capturing enhancement method and an audio capturing system using the same method. The audio capturing system includes at least but not limited to two microphones for recording an audio data, an amplifier coupled to the at least two microphones for adjusting the audio data by applying automatic gain control (AGC) in order to generate a gain adjusted data that is within a predefined level, and a processing circuit coupled to the amplifier for calculating a linear predictive coding (LPC) residue of the gain adjusted data, determining from the LPC residue a first source at a first direction relative to the at least two microphones based on time different of arrival (TDOA), and attenuating any source at a second direction that is outside of a predefined direction.

Abstract: Provided is a sound field security system with improved sound field analysis performance, and a method of determining a starting point for analysis of a received waveform by using the same, in which a starting point for analysis of a received waveform may be accurately determined even when there are irregular delays in a system, thereby enabling sound field analysis to be more adaptive to irregular delays that may occur in the system, and ensuring stable sound field security.

Abstract: A method executed by one or more electronic devices in an electronic system moves binaural sound of a voice of a person during a telephone call. During the telephone call, the voice of the person is convolved with head related transfer functions (HRTFs) to externally localize the voice as binaural sound. A command moves the voice from externally localizing as binaural sound to internally localizing as one of mono sound or stereo sound.

Abstract: A method executed by one or more electronic devices in an electronic system provides binaural sound of a voice of a person to localize at an image during a telephone call. During the telephone call, a wearable electronic device displays an image of a calling party, and a voice of the calling party externally localizes as the binaural sound to the image of the calling party.

Abstract: A digital microphone system, audio control device and control method thereof is related to the method for controlling a digital microphone circuit including receiving a clock signal; detecting that the clock signal is maintained at a predetermined level for a duration; when the duration reaches a given time, switching a transmission type of a data pin; in a data mode, outputting a digital audio signal to a data line via the data pin in an output type; and in a command mode, receiving a command signal from the data line via the data pin in an input type.

Abstract: 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.

Abstract: The present invention relates to an earset, comprising: a first earphone portion, which includes a first speaker for outputting sound signals or voice signals that are provided from an external device, and which can be inserted into a first external auditory canal of a user; a second earphone portion, which includes a first microphone for receiving inputted user voice signals that are provided through the external auditory canal of the user, and which can be inserted into a second auditory canal of the user; and a main body connected to each of the first and the second earphone portion. When the main body is wirelessly connected to the external device, the main body comprises; a signal transceiving portion for transceiving the signals with the external device; and a control portion for outputting via the first speaker the voice signals received from the external device through the signal transceiving portion.

Abstract: Filtering containing a delay by a specific time is performed on one of the pair of input signals are input from microphones L, R. After the filtering, a pair of input signals InL and InR are alternately interchanged for each sampling by an interchanging circuit 2 to generate a pair of interchanged signals InA and InB. The one interchanged signal InB is multiplied by a coefficient m by an coefficient updating circuit 3 to generate an error signal of the interchanged signals InA and InB. The recurrence formula of the coefficient m containing the error signal is calculated to update the coefficient m for each sampling. The pair of input signals InL and InR are multiplied by the sequentially updated coefficient m and are output.

Abstract: A camera is configured with multiple microphones to reduce wind noise in audio data collected by the camera. The camera receives motion data, which may comprise data indicating acceleration of the camera, a plurality of video frames received by the camera, or a background level of noise associated with one or more microphones configured on the camera. The camera determines a motion vector from the motion data. The motion vector is parallel to the direction of motion of the camera. The camera selects a subset of one or more microphones in the direction of the motion vector. By recording audio data using the one or more selected microphones, the camera reduces wind noise in the audio data.