Abstract: Examples disclosed herein provide the ability to identify the location of an individual within a room by using a combination of microphone arrays and voice pattern matching. In one example, a computing device may extract a voice detected by microphones of a microphone array located in a room, perform voice pattern matching to identify an individual associated with the extracted voice, and determine a location of the individual in the room based on an intensity of the voice detected individually by the microphones of the microphone array.

Abstract: Provided are, among other things, systems, methods and techniques for audio-signal processing. One representative embodiment includes HT sub-band analysis/decomposition modules, e.g., one for each audio channel and one for an echo reference signal. Each HT sub-band analysis/decomposition module includes a Hilbert Transformation module and an analysis/decomposition filter bank and provides sub-band outputs. Echo-cancellation modules, e.g., one for each audio channel, perform echo-cancellation processing on such sub-bands. Beamforming modules, e.g., one for each sub-band, then perform beamforming, e.g., across all audio channels. Finally, a resynthesis stage combines the different sub-band outputs in order to provide a system output signal.

Abstract: Embodiments provide techniques for altering a virtual world based on combinational input gestures. Embodiments retrieve a definition for a combinational gesture within a computer game, the definition specifying physical actions to perform according to a specified timing schedule in order to successfully perform the combinational gesture. User activity is monitored to detect when a first user input from a first input devices sufficiently matches a first predefined pattern of user input corresponding to a first physical action, and to detect when a second user input from a second input device sufficiently matches a second predefined pattern of user input corresponding to a second physical action. Embodiments determine that the first and second user inputs were performed according to the timing schedule specified in the definition. A status of at least one aspect of a virtual world for the computer game is altered, based on performance of the combinational gesture.

Abstract: A headphone, headphone system, and speech enhancing method is provided to enhance speech pick-up from the user of a headphone and includes receiving a plurality of signals from a set of microphones and generating a primary signal by array processing the microphone signals to steer a beam toward the user's mouth. A noise reference signal is also derived from one or more microphones, and a voice estimate signal is generated by filtering the primary signal to remove components that are correlated to the noise reference signal.

Abstract: An audio processing device includes first and second LPFs for outputting low-frequency components of a first right channel signal and a first left channel signal; a first subtracter for subtracting an output signal of the second LPF from the first right channel signal, thereby outputting a second right channel signal; a second subtracter for subtracting an output signal of the first LPF from the first left channel signal, thereby outputting a second left channel signal; a third LPF for outputting a low-frequency component of a signal obtained by addition of the first right channel signal and the first left channel signal; a first amplifier for amplifying an output signal of the third LPF; and adders for adding up the second right channel signal and an output signal of the first amplifier and to add up the second left channel signal and the output signal of the first amplifier.

Abstract: A wireless transceiver including: at least one antenna; a plurality of transmit and receive path components; an array of microphones; an acoustic spatial diagnostic circuit and a rule execution circuit. The plurality of components form transmit and receive paths coupled to the at least one antenna. The acoustic spatial diagnostic circuit couples to the array of microphones to successively sample an acoustic environment surrounding the wireless transceiver and to determine from each set of acoustic samples an acoustic spatial map of at least humans within the surrounding environment. The rule execution circuit couples to the spatial diagnostic circuit to execute an action proscribed by a selected rule when a related portion of the acoustic spatial map sampled by the spatial diagnostic circuit exhibits a correlation above a threshold amount with a spatial context condition associated with the selected rule.

Abstract: In accordance with embodiments of the present disclosure, a method for voice processing in an audio device having an array of a plurality of microphones wherein the array is capable of having a plurality of positional orientations relative to a user of the array, is provided. The method may include periodically computing a plurality of normalized cross-correlation functions, each cross-correlation function corresponding to a possible orientation of the array with respect to a desired source of speech, determining an orientation of the array relative to the desired source based on the plurality of normalized cross-correlation functions, detecting changes in the orientation based on the plurality of normalized cross-correlation functions, and responsive to a change in the orientation, dynamically modifying voice processing parameters of the audio device such that speech from the desired source is preserved while reducing interfering sounds.

Abstract: A headset with an electro-acoustic input transducer arranged to pick up an acoustic signal and convert the acoustic signal to an electric signal. Based on processing a portion of the electric signal, the voice activity detector is configured to: detect proximal voice activity, distal voice activity and no voice activity, at times when respectively present in the acoustic signal picked up by the electro-acoustic transducer, and to select a respective mode, the selection of which is encoded in the control signal. The first processor is controlled by the voice activity detector to reduce, in the output signal, intelligibility of distal voice activity at least at portions of time periods when the control signal indicates the mode of presence of distal voice activity.

Abstract: The technology described in this document can be embodied in an apparatus that includes an array of multiple microphones, and a passive directional acoustic element disposed between at least two of the multiple microphones. The passive directional acoustic element includes a pipe having an elongated opening along at least a portion of the length of the pipe, and an acoustically resistive material covering at least a portion of the elongated opening. One or more structural characteristics of the passive acoustic element is configured for capturing a target frequency range in accordance with a target beam pattern associated with the array.

Abstract: Technologies for localized audio enhancement of a three-dimensional video include capturing a three-dimensional video including depth data using a three-dimensional camera of a mobile computing device and receiving a three-dimensional audio associated with the three-dimensional video using a microphone array. A user may enhance localized audio by selecting a region of the three-dimensional image from the three-dimensional video. In response, the mobile computing device generates an audio component of the three-dimensional audio corresponding to the selected region based on depth data associated with the selected region. A user may subsequently enhance the audio component by, for example, increasing the volume of the audio component or increasing the clarity of the audio component.

Abstract: Methods and apparatus to improve the accuracy of crediting media exposure through detecting reverberation indicative of spillover are disclosed. An example apparatus includes a reverberation analyzer to identify a quantity of short durations of loudness in an audio signal of media presented by a media presentation device and calculate a ratio of the quantity of the short durations of loudness to a quantity of durations of loudness in the audio signal of the media, the quantity of the durations of loudness including the quantity of short durations of loudness. The example apparatus also includes a processor and memory in circuit with the processor, the memory including instructions that, when executed by the processor, cause the processor to mark the media as un-usable to credit a media exposure when the ratio does not satisfy a loudness ratio threshold.

Abstract: A sound source probing apparatus, including storage and processing circuitry, is provided that probes a direction of a sound source. The processing circuitry performs operations including determining a first correlation matrix that is a correlation matrix of acoustic signals acquired as observation signals by a microphone array including two or more microphones disposed apart from each other. The operations also include determining, by learning, weights such that a linear sum of a plurality of second correlation matrices multiplied by the respective weights is equal to the first correlation matrix where the plurality of second correlation matrices are correlation matrices, which are determined for respective directions determined based on an array arrangement of the microphone array and which are stored in advance in the storage.

Abstract: A sound collecting system includes a plurality of sound collecting units that collect sound and a sound collecting device. The plurality of sound collecting units send sound collection data including audio data and time data to the sound collecting device. The sound collecting device includes a processor that manages time for the plurality of sound collecting units and receives an instruction specifying a sound collecting location, and an output unit. The processor of the sound collecting device synthesizes the audio data of the plurality of sound collecting units on the basis of the time data of the plurality of sound collecting units, and outputs, from the output unit, the audio data of the sound collecting location.

Abstract: An apparatus is configured to rest on a wearer's neck or shoulders, and has a first microphone positioned at a first position located on the right side of the wearer, when the apparatus is worn, and a second microphone positioned at a second position located on the left side of the wearer. A processor receives audio signals output by the first and second microphones, and compares the audio signals from the first microphone to the audio signals from the second microphone. Based on the comparison, the processor determines whether the wearer's head is rotated to the right, left, or center, during an instance of the user speaking. Based on the determination, the processor selects a first one of at least two audio signal destinations, and provides the audio signal from at least one of the microphones to the first selected audio signal destination.

Abstract: A hearing system includes a pair of first and second hearing devices wirelessly coupled to a remote device that includes a microphone. One or more gains can each be calculated as a function of a first microphone signal received from the first hearing device, a second microphone signal received from the second hearing device, and a remote microphone signal received from the remote device. The function can be designed to improve speech intelligibility in a noisy environment. The one or more gains are applied to the first and second microphone signals to produce output sounds by the first and second hearing devices.

Abstract: A dual microphone near field voice enhancement arrangement in a motor vehicle includes a seat having a headrest with two opposite lateral sides. Each of two microphones is mounted on a respective one of the two opposite lateral sides of the headrest. Each microphone produces a respective microphone signal indicative of sounds within a passenger compartment. An electronic processor receives the microphone signals. The processor calculates a time delay between the microphone signals, and uses the calculated time delay to estimate amplitudes of the microphone signals. The processor applies a respective time delay to each of the microphone signals based on the calculated time delay to produce two time-aligned signals. The processor applies a respective gain to each of the time-aligned microphone signals based on the estimated amplitudes to produce two time-aligned and gain corrected signals. The processor sums the time-aligned and gain corrected signals.

Abstract: An embodiment of the technology provides a wireless in-ear utility device that rests in the user's ear canal away from the user's eardrum. Embodiments of the in-ear utility device can be configured in a variety of ways, including, but in no way limited to, a multi flexible personal, Streaming Music, and multi internal and external communication device via wireless communication.

Abstract: An example method of operation may include designating sub-regions which collectively provide a defined reception space, receiving audio signals at a controller from the microphone arrays in the defined reception space, configuring the controller with known locations of each of the microphone arrays, assigning each of the sub-regions to at least one of the microphone arrays based on the known locations, and creating beamform tracking configurations for each of the microphone arrays based on their assigned sub-regions.

Abstract: Placement of microphones and design of filters in a microphone network are solved simultaneously. Using filterbanks with multiple sub-channels for each microphone, the design of the filter response is solved simultaneously with placement. By using an objective function that penalizes the number of sub-channels in any solution, only some of many possible sub-channels and corresponding microphones and filters are selected while also solving for the filter responses for the selected sub-channels. For a given target location, the location of the microphones and the filter responses to beamform are optimized.

Abstract: A spatialized audio presentation system identifies an orientation of a virtual avatar, associated with a user engaged in a virtual experience, with respect to a virtual sound source within a virtual space. Within the virtual space, the virtual sound source generates a sound to be presented to the user while the user is engaged in the virtual experience. Based on the identified orientation of the virtual avatar, the system selects a head-related impulse response from a library of head-related impulse responses corresponding to different potential orientations of the virtual avatar with respect to the virtual sound source. The system then generates respective versions of the sound for presentation to the user at the left and right ears of the user by applying, respectively, a left-side component and a right-side component of the selected head-related impulse response to the sound. Corresponding methods are also disclosed.

Abstract: A method of detecting respective source locations of sounds in an audio signal. The method includes receiving the audio signal via a horizontal set of microphones and a vertical set of microphones. The respective source locations of the sounds in the audio signal are determined by analyzing the audio signal. Analysis is conducted with respect to the horizontal set of microphones and with respect to the vertical set of microphones, to determine a respective horizontal direction to the respective source locations of the sounds, and determine a respective vertical direction to the respective source locations of the sounds. The distance is calculated between the respective source locations of the sounds and the horizontal set of microphones and the vertical set of microphones.

Abstract: A method of audio source separation from audio content is disclosed. The method includes determining a spatial parameter of an audio source based on a linear combination characteristic of the audio source and an orthogonality characteristic of two or more audio sources to be separated in the audio content. The method also includes separating the audio source from the audio content based on the spatial parameter. Corresponding system and computer program product are also disclosed.

Abstract: A single-ended signal transmission system recovers a noise signal associated with a data input signal and uses the recovered noise signal to compensate for noise on the data input signal. The noise signal may be recovered from a noise reference signal line, or clock signal line, or a data signal line associated with a DC-balanced data input signal. The recovered noise signal may be represented as an analog signal or a digital signal. The recovered noise signal may be processed to compensate for DC offset and nonlinearities associated with one or more different input buffers. In one embodiment, the recovered noise signal includes frequency content substantially below a fundamental frequency for data transmission through the data input signal.

Abstract: Speech received from a microphone array is enhanced. In one example, a noise filtering system receives audio from the plurality of microphones, determines a beamformer output from the received audio, applies a first auto-regressive moving average smoothing filter to the beamformer output, determines noise estimates from the received audio, applies a second auto-regressive moving average smoothing filter to the noise estimates, and combines the first and second smoothing filter outputs to produce a power spectral density output of the received audio with reduced noise.

Abstract: Noise is suppressed from a microphone array by estimating a noise field isotropy. In some examples audio is received from a plurality of microphones. A power spectral density of a beamformer output is determined and a power spectral density of microphone noise differences is determined. A noise power spectral density is determined using a transfer function and the noise power spectral density is applied to the beamformer output power spectral density to produce a power spectral density output of the received audio with reduced noise.

Abstract: A signal processing method and device are provided. At least two channel sound signals are acquired, and a frequency-domain audio signal corresponding to each channel sound signal is acquired; beam forming output signals of a beam group corresponding to an audio signal of each frequency point are acquired; an output direction of the beam group is acquired; and time-domain sound signals output after beam forming in the output direction are acquired.

Abstract: The disclosure relates to an apparatus for manipulating an input audio signal associated to a spatial audio source within a spatial audio scenario, wherein the spatial audio source has a certain distance to a listener within the spatial audio scenario. The apparatus comprises an exciter adapted to manipulate the input audio signal to obtain an output audio signal, and a controller adapted to control parameters of the exciter for manipulating the input audio signal based on the certain distance.

Abstract: An omnidirectional acoustic sensor is provided. The omnidirectional acoustic sensor includes first, inner resonators connected to an inner circumference of a supporting frame and second, outer resonators connected to an outer circumference of the supporting frame. The first, inner resonators vibrate in a height direction of the supporting frame and the second, outer resonators vibrate in a lateral direction of the supporting frame.

Abstract: An example of an apparatus configured to be worn by a person who has an ear and an ear canal includes a first microphone adapted to be worn about the ear of the person, and a second microphone adapted to be worn at a different location than the first microphone. The apparatus includes a sound processor adapted to process signals from the first microphone to produce a processed sound signal, a receiver adapted to convert the processed sound signal into an audible signal to the wearer of the hearing assistance device, and a voice detector to detect the voice of the wearer. The voice detector includes an adaptive filter to receive signals from the first microphone and the second microphone.

Abstract: A first microphone is operated in a low power sensing mode, and a buffer at the first microphone is used to temporarily store at least some of the phrase. Subsequently the first microphone is deactivated, then the first microphone is re-activated to operate in normal operating mode where the buffer is no longer used to store the phrase. The first microphone forms first data that does not include the entire phrase. A second microphone is maintained in a deactivated mode until the trigger portion is detected in the first data, and when the trigger portion is detected, the second microphone is caused to operate in normal operating mode where no buffer is used. The second microphone forms second data that does not include the entire phrase. A first electronic representation of the phrase as received at the first microphone and a second electronic representation of the phrase as received at the second microphone are formed from selected portions of the first data and the second data.

Abstract: A head mounted display device is used by being mounted on a body of a user and includes an image display unit through which outside scenery is transmitted and which displays an image such that the image is visually recognizable together with the outside scenery. Further, the head mounted display device includes a right headphone and a left headphone outputting a sound. Further, the head mounted display device includes a target detection unit that detects a target of the user in the visual line direction; a distance detection unit that detects a distance between the detected target and the user; and an information output control unit that controls the output of the sound of the right headphone and the left headphone according to the detected distance.

Abstract: An audio system includes an external microphone to receive a near-end audio content and a loudspeaker transducer and a corresponding enclosure defining an acoustic chamber. An internal pressure-gradient microphone is positioned in the acoustic chamber to detect a radiated output from the loudspeaker transducer. The audio system also includes a processor and a memory having instructions that, when executed by the processor, cause the audio system to receive a near-end signal from the external microphone and a reference signal from the internal microphone. The instructions, when executed, further cause the processor to cause the audio system to filter the reference signal from the near-end signal to define a clean near-end signal, and to emit the clean near-end signal. Related principles are described by way of reference to method and apparatus examples.

Abstract: Techniques are provided for QR Decomposition (QRD) based minimum variance distortionless response (MVDR) adaptive beamforming. A methodology implementing the techniques according to an embodiment includes receiving signals from microphone array, identifying signal segments that include a combination of speech and noise, and identifying signal segments that include noise in the absence of speech. The method also includes calculating a QRD and an inverse QRD (IQRD) of the spatial covariance of the noise components. The method further includes estimating a relative transfer function (RTF) associated with the source of the speech, based on the noisy speech signal segments, the QRD, and the IQRD. The method further includes estimating a multichannel speech-presence-probability (SPP) on whitened input signals based on the IQRD. The method further includes calculating beamforming weights, for the microphone array, based on the RTF and the IQRD, to steer a beam in the direction associated with the speech source.

Abstract: The technology described in this document can be embodied in an apparatus that includes an array of multiple microphones, and a passive directional acoustic element disposed between at least two of the multiple microphones. The passive directional acoustic element includes a pipe having an elongated opening along at least a portion of the length of the pipe, and an acoustically resistive material covering at least a portion of the elongated opening. One or more structural characteristics of the passive acoustic element is configured for capturing a target frequency range in accordance with a target beam pattern associated with the array.

Abstract: A speech-processing apparatus includes: a representative transfer function estimation unit that uses a sound signal which is collected by using a microphone array of which the arrangement is unknown, which has a plurality of channels, and of which the number of sound sources is unknown and that estimates a transfer function with respect to a sound source.

Abstract: An audio system for a vehicle has at least one source of audio signals. A respective directional loudspeaker array is mounted at each seat position and coupled to the at least one source. The at least one source includes a microphone that detects speech from an occupant of the first seat position. Processing circuitry receives signals from the microphone that correspond to the detect speech and drives each second respective loudspeaker array at the other seat positions to radiate acoustic energy corresponding to the detected speech.

Abstract: To remove only noise components without removing desired signal components, a signal processing apparatus includes a noise decorrelator that removes noise signals having correlation between at least two input signals, in each of which a desired signal and a noise signal coexist, by receiving the at least two input signals from at least two channels, and a residual noise remover that removes residual noise included in an output signal of the noise decorrelator based on a phase difference between the output signal of the noise decorrelator and at least one input signal included in the at least two input signals.

Abstract: An apparatus comprising: an input configured to receive at least two microphone signals associated with at least one acoustic source; an audio source determiner configured to determine from at least part of the at least two microphone signals at least one audio source based on the at least one acoustic source; an audio source direction determiner configured to determine at least one direction associated with the determined at least one audio source; a calibrator configured to calibrate at least one of the at least two microphone signals based on the at least one direction.

Abstract: A depth processing system can employ stereo speakers to achieve immersive effects. The depth processing system can advantageously manipulate phase and/or amplitude information to render audio along a listener's median plane, thereby rendering audio along varying depths. In one embodiment, the depth processing system analyzes left and right stereo input signals to infer depth, which may change over time. The depth processing system can then vary the phase and/or amplitude decorrelation between the audio signals over time to enhance the sense of depth already present in the audio signals, thereby creating an immersive depth effect.

Abstract: The present disclosure is directed to a gain control system that receives a signal and outputs a modulated signal. The signal modulation is based on a gain that is based on a detected distance between the gain control system and a user. The distance is detected with a ranging sensor which generates ranging data for a field of view, and then the gain control system analyzes the ranging data to identify the user, with a higher signal gain for the user at a greater distance and a lower signal gain for the user at a lesser distance.

Abstract: A noise signal is estimated based on a captured audio signal captured from a sound capture unit. It is determined whether the estimated noise signal thus estimated is in a noiseless state. If it is determined that the estimated noise signal is in the noiseless state, the captured audio signal is analyzed as a target sound signal, and a characteristic obtained by the analysis is learned and modeled, thereby generating a target sound model.

Abstract: Techniques, described herein, may enable a computing device (e.g., a set-top-box, a user equipment (UE), etc.) with an array of sensors (e.g., microphones, antennas, etc.) to engage in beamforming in a manner that optimizes the signal strength of the target signal. For example, the computing device may create an optimized beam pattern by applying a pre-selected band width to a steering direction of the target signal. The beam width may include steering vectors at equal, angular distances from the steering direction, enabling steep transition slopes to be applied to the optimized beam pattern. In some implementations, the beam width and transition slopes may be standardized, such that the optimized beam pattern for any signal may be uniform, regardless of variables such as frequency and time.

Abstract: The present disclosure discloses a sound adjustment system and a sound adjustment method, relating to the field of sound processing technology, to address the problem that outside noise has an adverse impact on the user's viewing quality. The sound adjustment system comprises: a processor, a noise detection module, a position detection module, and a light detection module; the noise detection module is configured to detect noise that affects normal broadcast of a television, and send noise volume data acquired to the processor; the position detection module is configured to detect the distance between a noise source and the television, and send position data acquired to the processor; the light detection module is configured to detect whether there is a user in front of the television, and send light detection data acquired to the processor; the processor controls broadcasting volume and broadcasting progress of the television according to the noise volume data, the position data and the light detection data.

Abstract: An acoustic signal processing apparatus includes circuitry to generate, when a plurality of sound receivers receive sound from a plurality of examination directions in a space and outputs acoustic signals of a plurality of channels, an effective signal corresponding to sound coming from each one of the examination directions based on the acoustic signals of the plurality of channels for each one of the examination directions, calculate a feature for each one of the examination directions based on the effective signal generated for each one of the examination directions, and select a target direction from the plurality of examination directions in the space based on the feature calculated for each one of the examination directions.

Abstract: A method of performing noise reduction includes capturing a first audio signal at a first microphone of a first device. The method also includes receiving, at the first device, audio data representative of a second audio signal from a second device. The second audio signal is captured by a second microphone of the second device. The method further includes performing noise reduction on the first audio signal based at least in part on the audio data representative of the second audio signal.

Abstract: A hearing assistance system comprises a left ear device and a right ear device respectively configured to be worn by a wearer. One or more microphones are provided at each of the left and right ear devices. One or more positional sensors are configured to determine a three-dimensional position of the hearing assistance system in response to the wearer looking at a sound source in space. A user interface is configured to receive an input directly from the wearer. A memory is configured to store the three-dimensional position of the hearing assistance system in response to the received input. A processor is configured to adjust a directional polar pattern of the one or more microphones provided at one or both of the left and right ear devices in response to the stored three-dimensional position.

Abstract: A loudspeaker cabinet has a number of pairs of microphones, each pair includes the same internal microphone and a different external microphone. For each pair of microphones, a process (i) receives a first audio signal of sound captured by the internal microphone and a second audio signal of sound captured by the different external microphone, (ii) estimates, using first and second audio signals, a radiation impedance, and (iii) computes a detection value based on the radiation impedance in a frequency band. A difference between (i) a currently computed detection value associated with a given pair of microphones and (ii) a previously computed detection value associated with said given pair, is computed. The sound produced by the cabinet is adjusted, in response to the computed difference meeting a threshold. Other embodiments are also described and claimed.

Abstract: A method for auralizing a multi-microphone device. Path information for one or more sound paths using dimensions and room reflection coefficients of a simulated room for one of a plurality of microphones included in a multi-microphone device is determined. An array-related transfer functions (ARTFs) for the one of the plurality of microphones is retrieved. The auralized impulse response for the one of the plurality of microphones is generated based at least on the retrieved ARTFs and the determined path information.

Abstract: A method includes estimating a spatial coherence between a first diffuse sound portion in a first microphone signal and a second diffuse sound portion in a second microphone signal. The first microphone signal is captured by a first microphone and the second microphone signal is captured by a second microphone which is spaced apart from the first microphone in a known manner. The method further includes defining a linear constraint for filter coefficients of a diffuse sound filter, the linear constraint being based on the spatial coherence. The method also includes calculating at least one of signal statistics and noise statistics over the first microphone signal and the second microphone signal. The method also includes determining the filter coefficients of the diffuse sound filter by solving an optimization problem concerning at least one of the signal statistics and noise statistics while considering the linear constraint for the filter coefficients.

Abstract: To improve, when area sound pick-up is performed to collect sounds from a sound source in a target area, the sound quality of the collected sounds. The present invention relates to a sound pick-up apparatus that performs area sound pick-up. The sound pick-up apparatus calculates a sound volume level of a mixing signal to mix with a target area sound on the basis of power of estimated noise obtained by estimating background noise included in an input signal input from a microphone array, and power of a non-target area sound, adjusts a sound volume level of the input signal, and a sound volume level of the estimated noise to mix with the mixing signal on the basis of the sound volume level of the calculated mixing signal, and generates and outputs a mixed target area sound with which the input signal that is adjusted to have the calculated sound volume level and the estimated noise that is adjusted to have the calculated sound volume level are mixed.