Abstract: An audio processing device and method uses audio signals from a virtual rotating microphone for acoustic angle of arrival detection using a doppler effect technique.

Abstract: A piezoelectric microelectromechanical system microphone comprises a support substrate, a piezoelectric element configured to deform and generate an electrical potential responsive to impingement of sound waves on the piezoelectric element, the piezoelectric element attached to the support substrate about a perimeter of the piezoelectric element, a sensing electrode disposed on the piezoelectric element and configured to sense the electrical potential, and corrugations defined in the piezoelectric element about the perimeter of the piezoelectric element to release residual stress and improve sensitivity of the piezoelectric microelectromechanical system microphone.

Abstract: A hearing device adapted to be worn by a user and for picking up sound containing the user's own voice is provided. The hearing device includes an input unit having first and second input transducers for converting sound to first and second electric input signals, respectively, representing the sound. The hearing device is configured to provide that the first and second input transducers are located on the user so that the user experiences first and second, acoustically different, acoustic environments, respectively, when the user wears the hearing device. The first acoustic environment may be defined as an environment where the own voice signal primarily originates from vibrating parts of the bones and skin or tissue. The second acoustic environment may be defined as an environment where the own voice signal primarily originates from the user's mouth or nose and is transmitted through air from said mouth or nose to the second input transducer(s). A method of operating a hearing device is further disclosed.

Abstract: An audio renderer can have a machine learning model that jointly processes audio and visual information of an audiovisual recording. The audio renderer can generate output audio channels. Sounds captured in the audiovisual recording and present in the output audio channels are spatially mapped based on the joint processing of the audio and visual information by the machine learning model. Other aspects are described.

Abstract: A microphone device includes a substrate, a plurality of microphones, a front panel, and a dustproof member. The plurality of microphones are arranged on the substrate. The front panel is configured to cover the substrate. The dustproof member is arranged so as to overlap the plurality of microphones in a plan view from a front panel side, and provided on a member different from the front panel.

Abstract: In one embodiment, an audio system can generate parametrically formulated noise signals. According to an embodiment, an audio system can both determine the hearing ability of an individual and increase the hearing ability of the individual by using parametrically formulated noise. According to an embodiment, an audio system can generate parametrically formulated noise having a power spectrum amplitude that is a function of an individual's hearing threshold across a range of frequencies as measured using parametrically formulated noise test signals.

Abstract: Described embodiments generally relate to a signal processing device for on ear detection for an earbud. The device comprises a first microphone input for receiving a microphone signal from a first microphone, the first microphone being configured to be positioned within an ear of a user when the earbud is being worn; a second microphone input for receiving a microphone signal from a second microphone, the second microphone being configured to be positioned outside the ear of the user when the earbud is being worn; a signal generator configured to generate a signal for acoustic playback from a speaker configured to be positioned within the earbud; and a processor.

Abstract: The disclosure relates to a soundbar and a method for automatic surround pairing and calibration of a surround sound system. The soundbar includes two built-in microphones on the left and right respectively, which can be used for determining relative positions of left and right surround speakers. When the relative positions of the left and right speakers are not correct, configurations of left and right surround channels can be automatically swapped with each other without manually swapping physical positions of the surround speakers by a user. In addition, latencies including a latency of each channel of a main system may also be calibrated, and magnitude compensation may be achieved by calculating a filter compensation coefficient of each line and merging it into an original filter. The automatic surround pairing and calibration of the surround sound system may be one-click completed automatically by the user by pressing a start button.

Abstract: Microphones, microphone systems, and methods for capturing and processing sound are described. The microphones and microphone systems may adaptively change the direction from which sound is captured. The microphones and microphone systems avoid the need to provide arrays of microphones, while providing adaptive beamforming without a time delay between each channel of information, and multi-directional sound capture. A dependency between the frequency response and system size is also avoided.

Abstract: A method for directional signal processing for an acoustic system, wherein first and second input transducers generate a first and a second input signal from an ambient sound. First and second intermediate signals are generated from the first and second input signal, wherein a preliminary superposition parameter is obtained for a first superposition of the first intermediate signal and the second intermediate signal in such a way that for the first superposition an attenuation in a first target direction has a maximum. A superposition parameter is formed from the preliminary superposition parameter so that a second superposition of the first and second intermediate signals, formed in the first target direction with the superposition parameter, has a pre-specified first value for a gain that is greater than zero. An output signal of the acoustic system is formed from the second superposition.

Abstract: According to certain embodiments, a microphone array having a plurality of microphone elements is calibrated by ensonifying the microphone array at a first direction relative to the microphone array with a first acoustic signal to concurrently generate a first set of audio signals from two or more of the microphone elements and processing the first set of audio signals to calibrate the two or more microphone elements. One or more other sets of audio signals can be generated by ensonifying the microphone array with one or more other acoustic signals at one or more other directions relative to the microphone array, where the two or more microphone elements are calibrated using the first set and the one or more other sets of audio signals. The calibration process can be performed outside of an anechoic chamber using one or more acoustic sources located outside or inside the microphone array.

Abstract: An image capture device with dynamic wind noise compression tuning techniques is described. A technique includes detecting of the presence of wind noise by measuring coherence between at least two microphones. For a compressor, adjusting a default compression threshold and default compression parameters based on the coherence measurements. For each microphone, applying by the compressor the adjusted compression parameters when an audio signal is above the adjusted compression threshold and applying the default compression parameters when the audio signal is below the adjusted compression threshold.

Abstract: A microelectromechanical systems (MEMS) microphone and form-factor adapter can include an adapter housing including an opening and an outer acoustic port and can include a MEMS microphone disposed at least partially within the adapter housing. The MEMS microphone can include a microphone housing, a MEMS motor disposed in the microphone housing and acoustically coupled to the outer acoustic port of the adapter housing via an acoustic port of the microphone housing, and an electrical circuit disposed in the microphone housing and electrically coupled to the MEMS motor and to electrical contacts on an exterior of the microphone housing. The electrical contacts can be physically accessible through the opening of the adapter housing. The adapter housing can change a form-factor of the MEMS microphone.

Abstract: Many headsets include automatic noise cancellation (ANC) which dramatically reduces perceived background noise and improves user listening experience. Unfortunately, the voice microphones in these devices often capture ambient noise that the headsets output during phone calls or other communication sessions to other users. In response, many headsets and communication devices provide manual muting circuitry, but users frequently forget to turn the muting on and/or off creating further problems as they communicate. To address this, the present inventors devised, among other things, an exemplary headset that detects the absence or presence of user speech, automatically muting and unmuting the voice microphone without user intervention. Some embodiments leverage relationships between feedback and feedforward signals in ANC circuitry to detect user speech, avoiding the addition of extra hardware to the headset.

Abstract: A speech processing apparatus includes a plurality of microphones configured to receive a plurality of input signals, and processing circuitry configured to generate a spatial filtering signal corresponding to the plurality of input signals through spatial filtering, generate estimated noise information by integrating directional noise information representing a level of a noise signal received from a direction of interest with diffuse noise information representing levels of noise signals received from various directions based on whether the plurality of input signals have directionality, and generate an estimated speech signal by filtering the spatial filtering signal based on the estimated noise information.

Abstract: Systems and methods for two-way communication with a hearing device are disclosed. In one embodiment, an accessory for communication with a hearing device includes an acoustic filter, and a microphone configured as an acoustic receiver (RX) for acoustic signals from a speaker of the hearing device via the acoustic filter. The acoustic filter is configured to operate at at least one resonance frequency. The accessory is in acoustic and magnetic communication with the hearing device.

Abstract: There is provided an information processing device and an information processing method that enable speeding up of a responsivity of a system response to a speech of a user. The information processing device includes a processing unit configured to determine, on the basis of a result of semantic analysis that is to be obtained from an interim result of speech recognition of a speech of a user, presence or absence of a response to the speech of the user. It thereby becomes possible to speed up a responsivity of a system response to the speech of the user. The present technology can be applied to a speech dialogue system, for example.

Abstract: According to one embodiment, a system for voice assistant tracking and activation includes a tracking component, a wake component, a listening component, and a link component. The tracking component is configured to track availability of a plurality of voice assistant services. The wake component is configured to determine a plurality of wake words, each plurality of wake words corresponding to a specific voice assistant service of the plurality of voice assistant services. The listening component is configured to receive audio and detect a first wake word of the plurality of wake words that corresponds to a first voice assistant service of the plurality of voice assistant services. The link component is configured to establish a voice link with the first voice assistant service for voice input by the user.

Abstract: Various implementations include microphone arrays and related speaker systems. In one implementation, a microphone array is mounted in a housing having a primary X axis, a primary Y axis perpendicular to the primary X axis, and a primary Z axis perpendicular to the primary X axis and the primary Y axis. The microphone array can include: a set of microphones positioned in a single plane perpendicular to the primary Z axis of the housing and axially asymmetric with respect to both the primary X axis of the housing and the primary Y axis of the housing. In some cases, all microphones in the set of microphones are stationary relative to the housing.

Abstract: The present invention relates to a communication disruption system. In a first audio path, a microphone receives input sound, an amplifier system amplifies the sound, and a sound system transmits a first output sound. In a second audio path, the microphone receives input sound, a delay circuit delays the sound, the amplifier system amplifies the sound, and the sound system transmits a second output sound. A target speaker will hear the first and second output sounds, with the first output sound being a reproduction of their speech heard nearly simultaneously with the original speech, and the second output sound being a reproduction of their speech heard slightly after the original speech. Due to the delayed auditory feedback effect, the target speaker's concentration will be disrupted, making it difficult for them to continue speaking.

Abstract: In one example, a video endpoint obtains, from a vertical microphone array, a first audio signal including audio from a target sound source and audio from a horizontally-displaced sound source. The video endpoint obtains, from a horizontal microphone array, a second audio signal and a third audio signal both including the audio from the target sound source and the audio from the horizontally-displaced sound source. Based on the second audio signal and the third audio signal, the video endpoint determines at least one of a first degree of arrival of the audio from the target sound source or a second degree of arrival of the audio from the horizontally-displaced sound source. Based on the at least one of the first degree of arrival or the second degree of arrival, the video endpoint adjusts a gain of the first audio signal.

Abstract: A multistage noise shaping (CT-MASH) converter with phase alignment is provided. The CT-MASH converter may include a prefilter, an auxiliary path with an adjustable continuous time sigma delta converter (CTSD), and a modulator. The adjustable CTSD may provide phase alignment using one or more of a variety of techniques, such as modifying a group-delay of the CTSD by tuning a feedforward coefficient, by tuning an excess loop delay coefficient, and/or by adjusting a clock timing of the CTSD.

Abstract: A hearing aid device comprising a microphone, a processing unit and a receiver for compensating for a hearing loss in the auditory system of a human is disclosed. The hearing aid device is configured with a microphone inlet assembly which is optimized with regards to maintenance of the hearing aid and improved sound performance. That is, the microphone inlet assembly comprises an inlet element attached to a PCB and further configured to connect with a sound inlet structure.

Abstract: An equalizer of a noise-reduction apparatus amplifies a predetermined amplification frequency band of a sending sound signal so as to generate an adjustment sending sound signal. A noise estimator estimates a noise included in the adjustment sending sound signal generated by the equalizer so as to generate an estimated noise signal. A noise level adjuster adjusts a level of the estimated noise signal generated by the noise estimator, in a predetermined adjustment frequency band, so as to generate an adjusted noise signal. A noise reducer reduces the noise signal included in the adjustment sending sound signal generated by the equalizer, utilizing the adjusted noise signal generated by the noise level adjuster.

Abstract: Systems and methods for crowdsourced noise monitoring are provided. An example for crowdsourced noise monitoring includes a noise monitoring system receiving and storing audio files and context data associated with a noise event in a database. A log of the audio files and context data can be created. Noise complaints can also be received from users by the noise monitoring system. Additional logs of the noise complaints can then be created. The log of the noise event and the additional logs of the noise complaints are analyzed to determine whether the noise complaints are related to the noise event. If the noise complaints are related to the noise event, the log and additional logs can automatically be forwarded to an authorized user.

Abstract: A controller 102, 202 for controlling a plurality of light sources is disclosed. The controller 102, 202 comprises a receiver 106, 206 configured to receive a sound input from a plurality of microphones, the sound input being a user input from a user, a localization module 108, 208 configured to determine a user location of the user relative to the plurality of microphones based on differences between the sound input received at different microphones of the plurality of microphones, and a processor 110, 210 configured to receive location information indicative of locations of the plurality of light sources, determine which one or more light sources of the plurality of light sources are associated with the user location based on the location information, and control the one or more light sources based on the sound input.

Abstract: A system, method and computer-readable storage device are disclosed for managing a mute and unmute feature on a device which is used to communicate data in a communication conference. The method includes detecting, when the device is set to mute, whether the user is speaking and whether the speech is meant for the conference. Background noises are distinguished from the speech of the user. If the user is speaking and the device is set to mute, the device will automatically switch to and unmute setting such that people in the indication conference can hear the user speak. Facial recognition, and gaze detection or other data can also be used to determine when to automatically mute or unmute the device and can aid in inferring an intent of the user to speak to the conference participants.

Abstract: A transducer assembly including a transducer enclosure having an enclosure wall separating a surrounding ambient environment from an encased space, and a transducer module positioned within the encased space. The transducer module has a module wall that divides the encased space into an exterior chamber and an interior chamber and defines a fluid port between the exterior chamber and the interior chamber, the exterior chamber is between the module wall and the enclosure wall, the interior chamber is between the module wall and a sound radiating surface positioned within the transducer module, and the interior chamber is acoustically coupled to an acoustic port to the surrounding ambient environment. The enclosure wall is movable relative to the module wall and movement of the enclosure wall causes ejection of a fluid out of the interior chamber to the ambient environment.

Abstract: A camera system with image and audio capture capabilities is configured to protect the internal audio components from the external environment. The camera system includes an internal audio assembly with a microphone and an audio circuit board. The audio circuit board is structured such that a gap exists within the board that allows transmission of sound waves from external the environment to the microphone. The audio assembly also includes a compressible annulus, a support annulus, and a waterproof membrane coupled by a support structure. The waterproof membrane protects the internal components from moisture while still allowing transmission of sound waves. The support structure, compressible annulus, support annulus, and audio circuit board are structured such that a gap exists above the microphone and underneath the waterproof membrane.

Abstract: A speech-processing system can engage in time synchronization between audio capture devices so that if multiple devices detect and process a same utterance, the system may determine which device is closer to a speaker of the utterance by comparing time stamp data from the multiple devices. The devices may synchronize their individual clocks to some time standard that may be disseminated among the devices and/or to a server. When an utterance is captured, the capturing devices may create a time stamp corresponding to when the respective devices detected the utterance. The time stamps may be linked to the synchronized time standard or may be converted to the time standard by another device. The system may then compare the audio data for incoming utterances, and if the same utterance comes from multiple devices, may use the time stamp data to determine which device first detected the utterance.

Abstract: A hearing aid has a first input transducer that generates a first input signal from an ambient sound signal, a second input transducer that generates a second input signal from the sound signal and at least one output transducer. A first direction is assigned to a first useful signal source and a second direction is assigned to a second useful signal source, which is spatially separated from the first useful signal source. Based on the first input signal and the second input signal, a first reference signal oriented in the first direction and a second reference signal oriented in the second direction are formed. An output signal is formed on the basis of the first reference signal and the second reference signal, and the output signal is converted into a sound signal by the output transducer of the hearing aid.

Abstract: Aspect of the disclosure provide a packaging technique for making a directional microphone which employs mechanical structures to cancel undesired background noise to realize the directional function instead of an extra sensor required in electronic noise-cancelling techniques, thus reducing footprint and cost of a directional microphone. A directional microphone based on this technique can include an acoustic sensor and a housing enclosing the acoustic sensor. The acoustic sensor can include a sensing diaphragm, a cavity below the sensing diaphragm, and a first substrate. The directional microphone device can further include a channel with an inlet open at an edge of the first substrate and an outlet connected with the cavity. The housing can include a cover attached to a second substrate supporting the first substrate. The cover can include a first opening over the sensing diaphragm and a second opening at a side of the cover.

Abstract: Techniques and mechanisms to facilitate connection with one or more integrated circuit (IC) dies of a packaged device. In an embodiment, the packaged device includes a first substrate coupled to a first side of a package, and a second substrate coupled to a second side of the package opposite the first side. Circuitry, coupled via the first substrate to one or more IC dies disposed in the package, includes a circuit structure disposed at a cantilever portion of the first substrate. The cantilever portion extends past one or both of an edge of the first side and an edge of the second side. In another embodiment, a hardware interface disposed on the second substrate enables coupling of the packaged device to another device.

Abstract: Systems, devices, and techniques for directional display and audio broadcast are described. It is determined whether a spoken query includes an individual query, a group query, and a display request. For an individual query, an audio response is broadcast in a directional manner towards the source of the spoken query. For a group query, an audio response is broadcast in more than one direction, such as towards directions of sources in an environment. For a display request, an image (e.g., picture, video, web page, and the like) is projected on a surface that is detected so the image is viewable by one or more users, such as the source of the spoken query.

Abstract: Techniques for providing instruction to devices in the network are provided. For example, based at least in part on the list of currently active devices and the command, a controller can determine a sequence of IR signals associated with at least one currently active device and transmit, utilizing an IR transmitter incorporated with the controller, the sequence of IR signals to one or more of the currently active devices. In another example, an audio command and location information can be received and used to determine where a controller should transmit an IR signal that corresponds with the audio command, based at least in part on the location information.

Abstract: A method for operating a video display device according to an embodiment of the present invention includes: transmitting a voice acquisition command signal to at least one peripheral device connected to the video display device; receiving at least one voice signal for a user voice acquired by at least one peripheral device having received the voice acquisition command signal, and a voice signal for the user voice acquired by the video display device; comparing the plurality of acquired voice signals with each other; determining a voice signal subjected to voice recognition based on the comparison result; recognizing the user voice based on the determined voice signal; and performing a control operation corresponding to the recognized voice.

Abstract: A method and apparatus for speech recognition are provided. The method and the apparatus calculate signal to noise ratios (SNRs) of speech signals from a user received at speech recognition apparatuses. The method and the apparatus recognize a reference speech signal having a maximum SNR among the SNRs.

Abstract: A voice command recognition apparatus and method thereof are described. The voice command recognition apparatus includes audio sensors placed at different locations; a context determiner configured to determine user context based on a voice received at the audio sensors, wherein the context comprises a vocalization from a user. A command recognizer in the voice command recognition apparatus is configured to activate to recognize a voice command or remain inactive according to the recognized context.

Abstract: According to an in-vehicle control apparatus, it is configured to receive information of a plurality of operation keys displayed on a display, to identify from a result of voice recognition of a user's speech, the operation key corresponding thereto on the basis of the information of the operation keys, and to execute a function assigned to the operation key upon receiving an instruction for execution by the user using a decision key, so that it is possible to perform an operation without making contact directly with the plurality of operation keys displayed on the display and without the need of a complex key operation. Further, solely by mounting just one decision key on a steering wheel, it becomes possible to safely perform the operation about the operation key displayed on the display without the user releasing his/her hand from the steering wheel during driving.

Abstract: An estimated system gain spectrum of an acoustic system is generated, and updated in real-time to respond to changes in the acoustic system. Peak gains in the estimated system gain spectrum are tracked as the estimated system gain spectrum is updated. Based on the tracking, at least one frequency at which the estimated system gain spectrum is currently exhibiting a peak gain is identified. Based on the identification of the at least one frequency, an audio equalizer is controlled to apply, to a first speech containing signal to be played out via an audio output device of the audio device and/or to a second speech containing signal received via an audio input device of the audio device, an equalization filter to reduce the level of that signal at the identified frequency. The equalization filter is applied continuously throughout intervals of both speech activity and speech inactivity in that signal.

Abstract: The disclosed subject matter provides a main device and at least one secondary device. The at least one secondary device and the main device may operate in cooperation with one another and other networked components to provide improved performance, such as improved speech and other signal recognition operations. Using the improved recognition results, a higher probability of generating the proper commands to a controllable device is provided.

Abstract: Disclosed herein are speech recognition apparatuses, vehicles having the speech recognition apparatuses, and methods for controlling vehicles. According to an aspect, a speech recognition apparatus includes a speech input unit configured to receive a speech command from a user, a communication unit configured to receive the result of processing for speech recognition acquired by at least one user terminal located near the user, and a controller configured to compare the result of processing for speech recognition acquired from the speech command received by the speech input unit to the result of processing for speech recognition acquired by the at least one user terminal, thus processing the speech command according to the result of the comparison.

Abstract: An audio device that sound enhances, that physically supports the sound enhancing, and that has a capability to physically support a video recorder. The audio device includes a microphone head and a stand. The microphone head enhances sound. The stand physically supports the microphone head and has the capability to physically support the video recorder.

Abstract: An audio equipped fan is disclosed having a housing defining an inner cavity, a motor disposed at least partially in the inner cavity of the housing and having an output shaft extending therefrom that is rotatable by the motor, a fan connected to the output shaft of the motor and rotatable therewith, a grille connected to the housing and positioned in alignment with the fan, the grille having an interior side and an exterior side and defining first openings through which air may flow while the fan is rotated and second openings through which sound may travel, and having a speaker connected to at least one of the housing, motor, fan and grille and aligned on the interior side of the grille with the second openings of the grille so that sound may travel from the speaker through the grille. Related methods are also disclosed.

Abstract: A processing unit that adaptively suppresses wind noise in a hearing aid is provided. The processing unit (100) comprises a first microphone (105) and a second microphone (106). The analog signal from the first microphone is converted to a first digital signal (107) in a first A/D converter (113) and the analog signal from the second microphone is converted to a second digital signal (108) in a second A/D converter (114). The output of the first A/D converter is operationally connected to a first input of a subtraction node (111). The output of the second A/D converter is operationally connected to the input of an adaptive filter (109). The output of the adaptive filter (109) is branched and in a first branch operationally connected to the second input of the subtraction node (111) and in a second branch operationally connected to the input of the remaining signal processing in the hearing aid.

Abstract: A microphone module and method for suppressing feedback in a microphone. The module has a casing with a hollow bore therethrough and a microphone mounted in one end of the bore. The other end of the bore is completely covered by a film mounted onto the top of the casing. The film has at least one slit therethrough in the film portion that covers the other end of the bore. The method includes introducing a sound wave to a film having at least one slit therethrough that separates the film into at least two parts; generating a sound wave from each film part; and conveying the generated sound waves in a sound tube to a microphone as a rejoined sound wave.

Abstract: The present invention provides a sound-transmitting membrane for a microphone, to be disposed to a sound-collecting opening of the microphone so as to allow sound to transmit through the membrane while preventing a foreign matter from entering into the microphone through the opening. This membrane for a microphone prevents the entry of a foreign matter, such as fine dust and water vapor, that is difficult for conventional sound-transmitting membranes to block, and ensures the microphone performance. This membrane for a microphone is composed of a nonporous film or a multilayer membrane including a nonporous film, has a surface density of 30 g/m2 or less, and has a sound transmission loss of less than 3 dB in a frequency range of 300 to 4000 Hz.

Abstract: A system and method receives a plurality of data streams from a smart pen device and a computing device, and indexes the data streams to a synchronized time index. A processor receives a first data stream representing gesture data captured by a smart pen device, and a second data stream representing a sequence of states associated with applications executing on a computing device, such that each state identifies content displayed by a computing device while the gesture data is captured. For example, a state could be a particular page of a digital document displayed by the computing device. After receiving the first and second data streams, the processor indexes the data streams to a synchronized time index, and stores the indexed data streams in a memory.

Abstract: An active noise control (ANC) processor has an adaptive filter that uses a reference signal to produce an anti-noise signal, and an error signal to evaluate cancellation performance. An adaptive filter algorithm engine configures the filter coefficients of the adaptive filter, in accordance with pre-shaped versions of the error and reference signals. The pre-shaping filter has a high-pass transfer function and enables the adaptive algorithm engine to increase noise cancellation performance in a high frequency band during the presence of focused or narrow-band noise energy in a low frequency band. Other embodiments are also described and claimed.

Abstract: Methods and system are described for cancelling interference in a microphone system. A positive bias voltage is applied to a first microphone diaphragm and a negative bias voltage is applied to a second microphone diaphragm. The diaphragms are configured to exhibit substantially the same mechanical deflection in response to acoustic pressures received by the microphone system. A differential output signal is produced by combining a positively-biased output signal from the first microphone diaphragm and a negatively-biased output signal from the second microphone diaphragm. This combining cancels common-mode interferences that are exhibited in both the positively-biased output signal and the negatively-biased output signal.