Abstract: A detection and ranging apparatus includes a probe signal generating unit that generates a probe signal according to a first modulation method, a transmitting unit that transmits the generated probe signal, a receiving unit that receives a signal including at least one of a reflection signal and an interference signal according to a second modulation method, a demodulating unit that demodulates the reception signal according to a method corresponding to at least one of the first modulation method and the second modulation method, an interference signal detecting unit that detects the interference signal from the demodulated signal, an interference signal identifying unit that identifies specifications and a delay amount of the interference signal from the demodulated signal, and an interference signal removing unit that removes the interference signal within the reception signal by using the identified specifications and the identified delay amount of the interference signal.

Abstract: Devices, systems, and methods provide direction finding of an acoustic signal source with respect to a voice-controlled device. The direction can be found without using elevation data, instead determining the horizontal location based on power values of the received signal. A large number of candidate vectors having values for azimuth, elevation, and power may be generated by a steered response power algorithm. The large number of vectors is reduced to a small number of reference azimuths spanning an azimuth range by associating the vectors with the closest reference azimuth and then calculating an average and/or maximum power of the associated vectors at each reference azimuth. The reference azimuth with the highest average (or maximum) power may be set as the direction of the signal source. Alternatively, each reference azimuth having an average (or maximum) power exceeding a threshold may be considered a direction of one of multiple sources.

Abstract: Provided is a variable directivity electret condenser microphone that can simplify a circuit configuration, and outputs an audio signal in an unbalanced manner. Included are electrically independent first and second electret condenser microphone units in which first and second fixed electrodes are arranged back to back and facing each other in a mutually non-conductive state, and first and second diaphragms are arranged facing the first and second fixed electrodes with fixed intervals from the first and second fixed electrodes, respectively, a first impedance converter having an input terminal connected to the first fixed electrode, a DC cut capacitor selectively connected between an output terminal of the first impedance converter and an input terminal of the second impedance converter, and a directivity variable switche that can alternatively select a mode from at least a first directivity mode to a third directivity mode.

Abstract: A method for cancelling/reducing acoustic echoes in speech/audio signal enhancement processing comprises using a received reference signal to excite an adaptive filter wherein the output of the adaptive filter forms a replica signal of acoustic echo; an adaptation step size is controlled for updating the coefficients of the adaptive filter; the adaptation step size is initialized by using an open-loop approach and optimized by using a closed-loop approach; one of the most important parameters with the open-loop approach is an energy ratio between an energy of a returned echo signal in an input microphone signal and an energy of the received reference signal; one of the most important parameters with the closed-loop approach is a normalized correlation or a square of the normalized correlation between the input microphone signal and the replica signal of acoustic echo; the replica signal of acoustic echo is subtracted from the microphone input signal to suppress the acoustic echo in the microphone input signal

Abstract: A sound source-separating device includes a sound-collecting part, an imaging part, a sound signal-evaluating part, an image signal-evaluating part, a selection part that selects whether to estimate a sound source direction based on the first sound signal or the first image signal, a person position-estimating part that estimates a sound source direction using the first image signal, a sound source direction-estimating part that estimates a sound source direction, a sound source-separating part that extracts a second sound signal corresponding to the sound source direction from the first sound signal, an image-extracting part that extracts a second image signal of an area corresponding to the estimated sound source direction from the first image signal, and an image-combining part that changes a third image signal of an area other than the area for the second image signal and combines the third image signal with the second image signal.

Abstract: Various techniques for detecting are described herein. In one example, a method includes detecting a position of a computing device and selecting a plurality of microphones to detect audio data based on the position of the computing device. The method can also include calculating location data corresponding to the audio data, the location data indicating the location of a user and modifying a far field gain value based on the location data.

Abstract: The present invention discloses a method and device for self-adaptively eliminating noises. Said method comprises: filtering the signal received by a first microphone using a first filter, filtering the signal received by a second microphone using a second filter, and obtaining a signal with noises reduced by subtracting the filtered signals; wherein, in a noise segment, the coefficients of the first filter the second filter are updated respectively using the signal with noises reduced such that the noise component contained in the signal filtered by the first filter tends to be the same with the noise component contained in the signal filtered by the second filter; and in a noisy voice segment, the coefficients of the first filter and the second filter are remained unchanged respectively, the first filter and the second filter respectively use a coefficient updated in the noise segment last time to filter the signals received by the first microphone and the second microphone.

Abstract: A device having built-in digital data means is powered by an unlimited power source for a lamp-holder, LED bulb, or light device connected to unlimited power source by prongs or a base that can be inserted into a socket that would otherwise receiving a bulb. The device may take the form of a webcam having auto tracking functions and retractable prongs that plug directly into a wall outlet.

Abstract: A system and method for tracking and tracing motions of multiple incoherent sound sources and for visualizing the resultant overall sound pressure distribution in 3D space in real time are developed. This new system needs only four microphones (although more could be used) that can be mounted at any position so long as they are not placed on the same plane. A sample configuration is to mount three microphones on the y, z plane, while the 4th microphone on a plane perpendicular to the y, z plane. A processor receives signals from the microphones based on the signals received from noise sources in unknown locations, and the processor determines the locations of these sources and visualizes the resultant sound field in 3D space in real time. This system works for broadband, narrowband, tonal sound signals under transient and stationary conditions.

Abstract: A method for audio signal processing is described. The method includes decomposing a recorded auditory scene into a first category of localizable sources and a second category of ambient sound. The method also includes recording an indication of the directions of each of the localizable sources. The method may be performed with a device having a microphone array.

Abstract: Disclosed are a device and a method for tracking a sound source location by removing wind noise. The disclosed method for tracking a sound source by removing wind noise comprises the steps of: performing a Fourier transform for input signals of a time domain received by a plurality of microphones; detecting a first sound source section on the basis of an average power value of the Fourier-transformed input signals; detecting a second sound source section from which wind noise has been removed, on the basis of the first sound source section and a difference of power values of the input signals received by the microphones; and detecting a location of a sound source on the basis of the second sound source section and a phase difference of the input signals received by the plurality of microphones.

Abstract: Embodiments of the present invention provide a condenser microphone with a movable lining and a fixed lining. The movable lining and the fixed lining are respectively connected to a power supply and an amplifier. The movable lining includes graphene (G), formed at least as one graphene layer, while the fixed pad is formed as a fixed plate.

Abstract: A videoconferencing endpoint includes at least one processor a number of microphones and at least one camera. The endpoint can receive audio information and visual motion information during a teleconferencing session. The audio information includes one or more angles with respect to the microphone from a location of a teleconferencing session. The audio information is evaluated automatically to determine at least one candidate angle corresponding to a possible location of an active talker. The candidate angle can be analyzed further with respect to the motion information to determine whether the candidate angle correctly corresponds to person who is speaking during the teleconferencing session.

Abstract: An apparatus comprising 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 at least to perform processing at least one control parameter dependent on at least one sensor input parameter, processing at least one audio signal dependent on the processed at least one control parameter, and outputting the processed at least one audio signal.

Abstract: A method of noise suppression in a hearing aid system by providing an improved noise estimate derived from the difference of a first digital audio signal provided by a first input transducer and an adaptively filtered second digital audio signal provided by a second input transducer. The invention further provides a hearing aid (100, 200) and a hearing aid system (300 and 400) adapted for improving noise suppression in accordance with this method.

Abstract: Methods, systems, and computer-readable and executable instructions for spatial audio database based noise discrimination are described herein. For example, one or more embodiments include comparing a sound received from a plurality of microphones to a spatial audio database, discriminating a speech command and a background noise from the received sound based on the comparison to the spatial audio database, and determining an instruction based on the discriminated speech command.

Abstract: A sound pickup apparatus, which can control directionality and has a simple structure, and in which a plurality of microphones can be provided so as to be floated by using the small number of parts, is provided. Three support columns 13A to 13C having the same length are provided upright on a top surface of a housing 11 of a sound pickup apparatus 1. The support columns 13A to 13C are equidistantly arranged away from a center position of the housing 11 and equally spaced 120 degrees apart from each other. A frame 4 having microphone frames 14A to 14C in which microphones are fitted is arranged above the top surface of the housing 11. In addition, the microphone frames 14A to 14C are coupled to the support columns 13A to 13C of the housing 11 by elastic rubbers 15A to 15C in a state where tensile stress passes the center of the housing 11 and is generated in an outward direction.

Abstract: An audio device may use the audio detected at two opposite facing, front and rear omnidirectional microphones to determine the angular directional location of a user's voice while the device in speaker mode or audio command input mode. The angular directional location may be determined to be at front, side and rear locations of the device during the period of time by calculating an energy ratio of audio signals output by the front and rear microphones during the period. Comparing the ratio to experimental data for sound received from different directions around the device may provide the location of the user's voice. Based on the determination, audio beamforming input settings may be adjusted for user voice beamforming. As a result, the device can perform better beamforming to combine the signals captured by the microphones and generate a single output that isolates the user's voice from background noise.

Abstract: An acoustic camera for using a MEMS microphone array comprises: an acoustic sensor apparatus (30) comprising a print circuit board (20) on which the plural of MEMS microphone (10) are mounted, to send signals for the detected sound to a data collection unit (40); a data collection unit (40) connected to the acoustic sensor apparatus (30), which samples analog signals related to sound transmitted from the acoustic sensor apparatus (30) to transform into digital signals and transmit them to the central processing unit (40); a central processing unit (50) connected to the data collection unit (40), which calculates noise level based on digital signals related to sound transmitted from the data collection unit (40); and a display unit (60) which is connected to the central processing unit (50), which displays in color the noise level calculated at the central processing unit.

Abstract: A sound receiving device includes a microphone array, a plurality of time delay circuits and a sound-mixing output device. The microphone array includes a plurality of microphones longitudinally arranged along a straight line in order, an output terminal of each microphone is connected with a time delay circuit, and an output terminal of the time delay circuit is connected to an input terminal of the sound-mixing output device; and an i-th time delay circuit has a delay time Ti defined by adding a (n?i) times of unit time to a delay time of a last time delay circuit. The device can increase the output of the forward acoustic wave actuation, decrease the output of the oblique acoustic wave within a certain frequency bandwidth, and obtain nearly the same directional characteristic at the central frequency and adjacent frequencies.

Abstract: A method for recording and reconstructing a three-dimensional (3D) sound field, wherein a microphone array is established in a 3D sound field to track and locate sound sources in the 3D sound field and retrieve corresponding sound source signals. A plurality of control points is established inside an area where the 3D sound field is to be reconstructed. The control points are used to establish relational expressions of the sound source signals, the 3D sound field, a reconstructed sound field, and reconstructed sound source signals. The reconstructed sound source signals are obtained via solving the relational expressions and input into a speaker array arranged outside the area to establish the reconstructed sound field in the area. The present invention truly records the 3D sound field without using any extra transformation process and replays the reconstructed sound field with a larger sweet spot in higher fidelity.

Abstract: A method for source selective real-time monitoring and mapping of environmental noise, comprising the steps of ? identifying sources in the area; ? designing places for the monitoring stations; ? creating a data collecting and data processing center; ? measuring the noise impact, and observation of the sources with the sensors; ? defining the noise propagation; ? assemblying the monitoring stations including a sound measuring device, a computing unit and a communication unit; ? determining the resultant noise impact prevailing in the respective measurement points; ? determining the sources dominant in the individual moments on the basis of the data of the sensors; ? determining the average noise impact of the respective sources for a longer period; ? obtaining an average noise impact; ? extending the measurement into a noise impact map; ? calculating the effective noise impact of every noise source in every point of the area for every period; ? producing the effective noise impact map of the respective period

Abstract: The present invention relates to a sound emitting and collecting apparatus including a sound collecting portion that captures surrounding sound using two microphones, and a sound emitting portion that emits sound from at least one speaker. The apparatus includes a sound source separating portion that extracts a target sound from a sound source in a predetermined direction, based on an input sound signal obtained by capturing surrounding sound using the two microphones, and an emission non-target sound removing portion that removes a non-target sound that is emitted from the speaker and captured by each of the microphones, based on sound source data for the sound emitting portion. The emission non-target sound removing portion is provided on a path that reaches the sound source separating portion. The emission non-target sound removing portion has a structure similar to an acoustic echo canceller, for example.

Abstract: A portable voice capture device having an orientable arm comprising a first leg and a second leg, the first leg and the second leg having different orientations, the first leg of the first arm comprising a first differential linear array of microphones, a directivity of the first differential linear array being arranged for improved sensing of voice from a user, the second leg of the arm comprising a second differential linear array of microphones, a directivity of the second differential linear array being arranged for improved sensing of noise from a different than the direction of said sensed voice, and a noise reduction circuit for providing a voice signal with reduced noise, based on the output of the first array and on the output of the second array.

Abstract: An the acoustic apparatus comprising a first MEMS motor that includes a first diaphragm and a first back plate, and a second MEMS motor that includes a second diaphragm and a second back plate. The first motor is biased with a first electrical polarity and a second motor is biased with a second electrical polarity such that the first electrical polarity and the second electrical polarity are opposite. At the first motor, a first signal is created that is representative of received sound energy. At the second motor, a second signal is created that is representative of the received sound energy. A differential output signal that is the representative of the difference between the first signal and the second signal is obtained. In obtaining the differential output signal, common mode noise between the first motor and the second motor is rejected.

Abstract: Spherical microphone arrays capture a three-dimensional sound field (P(?ct)) for generating an Ambisonics representation (Anm(t)), 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 equalization 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 present principles minimize that noise by using a Wiener filter processing (34) in the frequency domain, which processing is automatically controlled (33) per wave number by the signal-to-noise ratio of the microphone array.

Abstract: A beam-forming device includes a first target sound blocker 103 and a second target sound blocker 104 that remove a target signal having a correlation mutually from a first sound signal x1 and a second sound signal x2 which are converted by first and second microphones 101 and 102, a phase synchronizer 105 that synchronizes the phases of the first sound signal x1 and the second sound signal x2 and synthesizes these sound signals by using information acquired when the first target sound blocker 103 removes the target signal, and a noise learner 106 that learns a noise component included in an output signal of the phase synchronizer 105 from signals from which the target signal is removed by the first target sound blocker 103 and the second target sound blocker 104.

Abstract: Provided are methods and systems for spatially selecting acoustic sources using a post-processor that consists of a selection of one postfilter from a set of postfilters, or a cascade of postfilters, where each postfilter is optimal for a particular scenario. Each postfilter individually is based on optimizing the gain for each time-frequency bin based on knowledge of (i) a spatial covariance matrix for the desired source, (ii) a spatial covariance matrix for the interfering sources, and (iii) microphone signals in some neighborhood of the current time-frequency bin.

Abstract: A device and a method to improve quality of a signal in a lossy communication system are disclosed. One or more samples of the signal are received from a first and second microphone transducers. The received samples are processed and filtered to obtain a processed signal. A voice activity detector is provided for iteratively identifying speech regions and non-speech regions of the signal. All samples received by the microphones are continuously monitored and quality of each sample is improved by reducing or eliminating the noise detected in the non-speech regions of the processed signal.

Abstract: A method of manufacturing a microphone, a microphone, and a method of controlling the microphone are provided. The method includes forming a sound sensing module on a mainboard having a first sound aperture, to be connected with the first sound aperature and forming a cover having a second sound aperature that corresponds to the first sound aperature, mounted on the mainboard, and housing the sound sensing module. A first and second sound delay filters are formed in a space defined by the cover, to be connected with the second sound hole and thermal actuators are disposed at both sides of the first sound delay filter and move the first sound delay filter based on whether power is supplied. A semiconductor chip is electrically connected with the sound sensing module in the space and selectively operates the thermal actuators in response to signals from the sound sensing module.

Abstract: Configuring an adaptive microphone array to gather signals from a main lobe of the array, and configuring the array to reduce side interference gathered from sources that are not situated within the main lobe. A memory stores test signals gathered by the array at a plurality of predetermined angular bearings with reference to the array in an anechoic chamber. Signals gathered in real time are processed to provide a preliminary output and preliminary weights. The test signals are retrieved from memory. The preliminary weights are applied to the test signals to provide null steering weights. The null steering weights and the preliminary output are processed to reduce or minimize the amplitude response of the array at the angular orientation.

Abstract: A device is configured for identifying a direction of a sound. The device includes a controller comprising circuitry. The circuitry is configured to receive a first output from a first input device and a second output from a second input device. The circuitry is also configured to add a delay to the second output. The circuitry is also configured to compare the first output to the delayed second output in a plurality of directions to form a comparison. The circuitry is also configured to identify a number of null directions of the plurality of directions where a set of nulls exists based on the comparison.

Abstract: 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.

Abstract: A microphone array system for sound acquisition from multiple sound sources in a reception space surrounding a microphone array that is interfaced with a beamformer module is disclosed. The microphone array includes microphone transducers that are arranged relative to each other in N-fold rotationally symmetry, and the beamformer includes beamformer weights that are associated with one of a plurality of spatial reception sectors corresponding to the N-fold rotational symmetry of the microphone array. Microphone indexes of the microphone transducers are arithmetically displaceable angularly about the vertical axis during a process cycle, so that a same set of beamformer weights is used selectively for calculating a beamformer output signal associated with any one of the spatial reception sectors. A sound source location module is also disclosed that includes a modified steered power response sound source location method. A post filter module for a microphone array system is also disclosed.

Abstract: An acoustic system includes first one or mole acoustic elements designed and arranged in a first manner to facilitate generation of a first signal that includes mostly undesired audio, substantially void of desired audio, in response to a presence of the desired audio and the undesired audio. Second one or more acoustic elements are designed and arranged in a second complementary manner to facilitate generation of a second signal that includes both the desired and the undesired audio, in response to the presence of the desired audio and the undesired audio. A signal extraction component receives the first signal and the second signal. The signal extraction component further includes an inhibit component. The inhibit component is coupled to the first signal and the second signal. A delay element is coupled to a path of the second signal. The delay element introduces a deterministic delay to the second signal.

Abstract: A microphone array-based audio system that supports representations of auditory scenes using second-order (or higher) harmonic expansions based on the audio signals generated by the microphone array. In one embodiment, a plurality of audio sensors are mounted on the surface of an acoustically rigid polyhedron that approximates a sphere. The number and location of the audio sensors on the polyhedron are designed to enable the audio signals generated by those sensors to be decomposed into a set of eigenbeams having at least one eigenbeam of order two (or higher). Beamforming (e.g., steering, weighting, and summing) can then be applied to the resulting eigenbeam outputs to generate one or more channels of audio signals that can be utilized to accurately render an auditory scene.

Abstract: Systems and methods of improved noise reduction using direction of arrival information include: receiving audio signals from two or more acoustic sensors; applying a beamformer module to the audio signals to employ a first noise cancellation algorithm to the audio signals and combine the audio signals into an audio signal; applying a noise reduction post-filter module to the audio signal, the application of which includes: estimating a current noise spectrum of the audio signals after the application of the first noise cancellation algorithm; using spatial information derived from the audio signals received from the two or more acoustic sensors to determine a measured direction-of-arrival by estimating the current time-delay between the acoustic sensor inputs; comparing the measured direction-of-arrival to a target direction-of-arrival; applying a second noise reduction algorithm to the audio signal; and outputting a single audio stream with reduced background noise.

Abstract: A noise reduction system includes multiple transducers that generate time domain signals. A transforming device transforms the time domain signals into frequency domain signals. A signal mixing device mixes the frequency domain signals according to a mixing ratio. Frequency domain signals are rotated in phase to generate phase rotated signals. A post-processing device attenuates portions of the output based on coherence levels of the signals.

Abstract: A microphone array arrangement structure in a vehicle cabin is provided with multiple microphones arranged in a convex line. The microphones include a first microphone, and the direction of the axis of the first microphone faces the mouth of the driver. The microphone array arrangement structure is provided with covers housing the microphones and having multiple holes. The front surface of the covers form a convexly curved line, and each of the microphones is fixed inside of a cover to correspond to one of the holes.

Abstract: Spherical microphone arrays capture a three-dimensional sound field (P(?c,t) for generating an Ambisonics representation (Anm(t)), 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 equalization 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.

Abstract: Methods and apparatus to beamform a first plurality of microphone signals using at least one beamforming weight to obtain a first beamformed signal, beamform a second plurality of microphone signals using the at least one beamforming weight to obtain a second beamformed signal, and adjust the at least one beamforming weight so that the power density of at least one perturbation component present in the first or the second plurality of microphone signals is reduced.

Abstract: An apparatus and method for calibrating gain difference between microphones included in a microphone array are provided. In the gain calibrating apparatus, weights for each frequency component of the acoustic signals, which have been converted into the signals in the frequency domain are calculated. The weights are used to calibrate the acoustic signals such that the plurality of acoustic signals each have the same amplitude while the acoustic signals maintain their individual phase. The amplitudes of the acoustic signals are calibrated by use of the calculated weights. The gain calibrating apparatus calibrates gain in real time while calculating weights for frequency components of the frame of acoustic signals in real time.

Abstract: A microphone system has an output and at least a first transducer with a first dynamic range, a second transducer with a second dynamic range different than the first dynamic range, and coupling system to selectively couple the output of one of the first transducer or the second transducer to the system output, depending on the magnitude of the input sound signal, to produce a system with a dynamic range greater than the dynamic range of either individual transducer. A method of operating a microphone system includes detecting whether a transducer output crosses a threshold, and if so then selectively coupling another transducer's output to the system output. The threshold may change as a function of which transducer is coupled to the system output. The system and methods may also combine the outputs of more than one transducer in a weighted sum during transition from one transducer output to another, as a function of time or as a function of the amplitude of the incident audio signal.

Abstract: A directional characteristic of a microphone facility of a hearing system is more reliably controlled. The method determines a first feature value in respect of speech in a first signal of a microphone facility assigned to a first direction and a second feature value in respect of speech in a second signal of the microphone facility assigned to a second direction. A control value is obtained from the difference of the two feature values. The directional characteristic of the microphone facility is controlled with this control value.

Abstract: An apparatus for generating an audio output signal to simulate a recording of a virtual microphone at a configurable virtual position in an environment includes a sound events position estimator and an information computation module. The former is adapted to estimate a sound source position indicating a position of a sound source in the environment, wherein the sound events position estimator is adapted to estimate the sound source position based on first and second direction information provided by first and second real spatial microphones, respectively, located at first and second real microphone positions in the environment, respectively. The information computation module is adapted to generate the audio output signal based on a first recorded audio input signal, on the first real microphone position, on the virtual position of the virtual microphone, and on the sound source position.

Abstract: An audio-based system may perform audio beamforming and/or sound source localization based on multiple input microphone signals. Each input microphone signal can be calibrated to a reference based on the energy of the microphone signal in comparison to an energy indicated by the reference. Specifically, respective gains may be applied to each input microphone signal, wherein each gain is calculated as a ratio of a energy reference to the energy of the input microphone signal.

Abstract: A planar antenna array and articles of manufacture using the same are disclosed. In one embodiment, close-packed antenna elements, disposed on a substrate, number N where N=3x and x is a positive integer. Each of the close-packed antenna elements includes a substantially continuous photonic transducer arranged as an outwardly expanding generally logarithmic spiral having six turns. Each of the outwardly expanding generally logarithmic spirals may be a golden spiral. As an article of manufacture, the planar antenna array may be incorporated into a chip, such as a cell phone, or an article of clothing, for example.

Abstract: A system and method for controlling and adjusting a low-frequency response of a MEMS microphone. The system comprising the MEMS microphone, a controller, and a memory. The MEMS microphone includes a membrane and a plurality of air vents. The membrane configured such that acoustic pressures acting on the membrane cause movement of the membrane. The plurality of air vents are positioned proximate to the membrane. Each air vent of the plurality of air vents are configured to be selectively positioned in an open position or a closed position. The controller determines an integer number of air vents to be placed in the closed positioned, and generate a signal that causes the integer number of air vents to be placed in the closed position and causes any remaining air vents to be placed in the open position.

Abstract: Extending a microphone interface. One microphone interface extension includes a controller, a parent microphone, and a child microphone. The controller outputs a controller clock signal. The parent microphone receives the controller clock signal and generates a first data signal. The child microphone generates a second data signal and outputs the second data signal to the first parent microphone. The parent microphone receives the second data signal from the child microphone and outputs a combined data signal to the controller based on the first data signal and the second data signal. The parent microphone outputs the combined data signal to the controller on a phase of a microphone clock signal derived from the controller clock signal.

Abstract: Systems, methods, and devices for determining the location of one or more mobile devices within a vehicle comprising: (a) a controller located within the vehicle and configured to transmit at least two audio signals, a first audio signal directed generally into a driver space within the vehicle and a second audio signal directed generally into a passenger space within the vehicle, and (b) software code stored in memory of the mobile device and having instructions executable by a processor that performs the steps of: (i) detecting the at least two audio signals, (ii) sampling the at least two audio signals for a predetermined period of time; (iii) performing digital signal processing on the sampled at least two audio signals; and (iv) based on the results of the digital signal processing, determining whether the mobile device was located within the driver space of the vehicle during the predetermined period of time.