171results about "Microphone spatial/constructional arrangements" patented technology

A first electrical signal is received from a first cardioid microphone capsule of a microphone assembly. The first cardioid microphone capsule is oriented to face in a first direction. A second electrical signal is received from a second cardioid microphone capsule of the microphone assembly. The second microphone capsule is oriented to face in a second direction that is opposite the first direction. A polar pattern is formed for the microphone assembly by converting the first electrical signal into a first audio signal and converting the second electrical signal into a second audio signal. A first weight is applied to the first audio signal and a second weight is applied to the second audio signal. The polar pattern is generated for the microphone assembly by summing the first audio signal and the second audio signal.

The present disclosure provides an audio system, method, and system for facilitating machine operation. The machine includes an actuator that assists the tool in performing the task. In an example, an audio system is configured to receive an audio mixed signal of a signal generated by an audio source that includes at least one of a tool or an actuator that is performing a task. An audio source forming an audio mixed signal is identified by a relative position to each microphone of a microphone array that measures the audio mixed signal. The audio system is configured to extract an audio signal from an audio mixed signal generated by the identified audio source based on a correlation of spectral features in a multi-channel spectrogram of the audio mixed signal and directional information indicative of a relative position of the identified audio source. The audio system outputs the extracted audio signal to facilitate operation of the machine.

An audio system is provided. The audio system includes a plurality of loudspeakers and a mobile device. The plurality of loudspeakers is capable of being positioned in a listening environment and being arranged to transmit an audio signal in the listening environment. Each loudspeaker is programmed to determine angle information of the audio signal as received at the loudspeaker relative to other loudspeakers of the plurality of loudspeakers and to transmit a first signal indicative of the angle information. The mobile device is programmed to receive the first signal from each of the loudspeakers and to determine a location for each loudspeaker in the listening environment based at least on the angle information.

Techniques are described for wirelessly receiving, at first and second speaker assemblies, signals representing left and right audio channels from each of first and second sources, and using the signals, simultaneously playing, on both speaker assemblies, music, computer game audio, computer chat audio, and speech picked up by speaker microphones.

Example technologies described herein relate to spatial audio on wearable playback devices, such as headphone and earbuds. Such technologies may include forms of binaural rendering of audio, which is played back on the wearable playback devices. These technologies may create or enhance an immersive listening experience which, to the listener, does not sound like you are listening on a wearable device, but are instead listening to a well-tuned, well-place, three-dimensional audio system of loudspeakers in a particular listening environment, such as the listener's actual living room.

A lavalier microphone has improved dynamic range. The lavalier microphone incorporates a pair of microphone transducer elements into the same housing or attached housings, with one of the microphone transducer elements being more sensitive to acoustic input than the other. The voltage signals from the respective transducer elements are processed and combined, resulting in a digital output audio signal with improved signal to noise both at lower and higher input acoustic levels.

The present invention relates to a vehicle with a beamforming system for improving speech recognition in the interior of the vehicle and to a method for improving speech recognition in the interior of a vehicle. The beamforming system comprises a microphone array, wherein the microphone array includes a plurality of microphones arranged at known positions in the interior for capturing acoustic signals, wherein the beamforming system includes a camera system, wherein the camera system includes at least one camera configured to optically capture the interior, wherein an evaluation unit of the camera system is configured to determine the position of a speaking person in the interior, wherein the beamforming system is configured to calculate, based on the determined position of the speaking person, expected values ​​for time-of-arrival differences of a speech signal emitted by the speaking person for the plurality of microphones, wherein the beamforming system is configured toBased on the expected values ​​for the time-of-flight differences, the speech signal emitted by the speaker is determined from the acoustic signals captured by the microphones.

A display device includes: a display panel comprising a first area and a second area; a sound generator disposed on the display panel in at least one of the first area and the second area; a dividing member disposed on the display panel at the boundary between the first area and the second area and extending in a predetermined direction; and a blocking member disposed on the display panel in at least one of the first area and the second area and spaced apart from the dividing member.

Apparatus comprising one or more processors configured to: receive at least two microphone audio signals (101) for audio signal processing wherein the audio signal processing comprises at least a spatial audio signal processing (303) and beamforming processing (305); determine spatial information (304) based on the audio signal processing associated with the at least two microphone audio signals; determine focus information (308) for the beamforming processing associated with the at least two microphone audio signals; and apply a spatial filter (307) in order to synthesize at least one spatially processed audio signal (312) based on the at least one beamformed audio signal from the at least two microphone audio signals (101), the spatial information (304) and the focus information (308) in such a way that the spatial filter (307), the at least one beamformed audio signal (306), the spatial information (304) and the focus information (308) are configured to be used to spatially synthesize (307) the at least one spatially processed audio signal (312).

The invention relates to a method and an apparatus for improving the intelligibility of a group of persons engaged in conversation, each person in the group being able at times to be a speaker and at times to be a listener. The persons are situated at two or more positions P. For this purpose, a system (1) is used comprising: two or more directional microphones M which are directed towards the positions P; two or more directional loudspeakers L which are likewise directed towards the positions P; and a digital signal processor DSP which is capable of processing and forwarding acoustic signals in real time. Each microphone M continuously receives acoustic signals Ai and forwards them to the DSP. The DSP identifies, in real time, a speech signal Si1 from each acoustic signal Ai and generates therefrom a processed speech signal aSi1. The DSP transmits the processed speech signal aSi1 to one or more other loudspeakers L. The DSP detects later-arriving speech signals Si2, Si3 having the same characteristics as reflections of the speech signal Si1, and generates therefrom, in real time, processed reflected signals aSi2, aSi3, which the DSP transmits to another loudspeaker L within a predetermined time window (4) having a duration of at most 40 ms.

Embodiments described herein involve an auxiliary zone contributing audio to a primary zone. In an example implementation, a network media system determines that a first zone in the network media system is playing back a first type of audio content and that a second zone in the network media system is not playing back audio content. While the first zone is playing back the first type of audio content and the second zone is not playing back audio content, the network media system forms a temporarily playback configuration. In the temporary playback configuration, the first zone plays back primary audio content including full frequency range audio content and the second playback device of the second zone plays back auxiliary audio content including low frequency range audio content.

An apparatus, method, and computer program product for receiving spatial audio information captured by a plurality of microphones, receiving captured audio objects from an audio device wirelessly connected to the apparatus, determining an audio audibility value related to the audio device, determining whether the audio audibility value satisfies at least one criterion, and activating inclusion of audio objects captured by the audio device in the spatial audio information captured by the plurality of microphones in response to determining that the audio audibility value satisfies the at least one criterion.

A microphone assembly includes a microphone array and a mixer. The microphone array includes first, second, and third microphones. The mixer includes an analog switch assembly and an amplifier assembly configured to mix recording signals of the microphones in different manners to realize different recording patterns of the microphone array. The different recording patterns realizable by the microphone array may include the cardioid pattern, the omnidirectional pattern, the stereo pattern, and the bidirectional pattern.

To provide an information processing method which allows a user to accurately measure a head transfer characteristic of the user himself / herself.SOLUTION: There is provided an information processing method including a step in which an information processing device controls a user interface regarding measurement of a head transfer function of the user. In the information processing method, as control of the user interface, the information processing device controls output of guide information for guiding a direction of the head of the user to a designated direction.SELECTED DRAWING: Figure 9

An audio processing system may be configured to play a sound source through speakers of the audio processing system. The system may determine a loudness of the sound source as heard by a user of the electronic device. The system may determine a playback loudness for the sound source that matches the loudness of the sound source as heard by the user. The system may author an audio work which includes the sound source. In a playback environment, the sound source is output by speakers of a headworn device at the playback loudness.

[Object] To provide a mechanism that allows the quality of binaural reproduction to further increase. [Solution] A signal processing system including a first control unit configured to calculate a transfer characteristic corresponding to a difference between a first audio signal and a second audio signal that is acquired by a first acquisition unit, the first acquisition unit being configured to acquire an audio signal, the second audio signal corresponding to the first audio signal that is reproduced by a first reproduction unit, the first reproduction unit being configured to reproduce an audio signal, and generate a fourth audio signal by convolving an inverse characteristic of the calculated transfer characteristic with a third audio signal that is acquired by a second acquisition unit.

Methods and apparatuses for capturing and encoding ambisonic audio are described herein. An example ambisonic microphone may comprise multiple sets of microphone capsules arrayed around a central axis. Each set may include two microphone capsules positioned along a first axis that is perpendicular to the central axis and two microphone capsules positioned along a second axis that is perpendicular to the central axis and spaced apart from the first axis by a predetermined distance. The multiple sets may be arranged in a sequence such that each set of the multiple sets is rotated about the central axis in the same angular direction by a predetermined angle between adjacent sets in the sequence.

Disclosed are an audio playing and collection method, apparatus and device and a readable storage medium, relating to the field of audio processing. The method is applied to a terminal connected to a Bluetooth earphone. The terminal is configured with a terminal microphone, and the Bluetooth earphone is configured with an earphone microphone. The method comprises: displaying a program interface, and playing, through a Bluetooth earphone, first audio corresponding to the program interface in the form of stereo sound effect; receiving a voice input signal; and according to the voice input signal, notifying a terminal microphone to open a voice input channel so as to collect second audio as voice input content. When the first audio is played, through the Bluetooth earphone, in the form of stereo sound effect, the second audio is collected by means of the terminal microphone to serve as the voice input content, the collection of the second audio is realized by means of the terminal microphone, while ensuring that the first audio played through the Bluetooth earphone is audio played in the form of stereo sound effect, and avoiding the problem that the audio cannot be collected while a stereo sound is played through the Bluetooth earphone.

The invention relates to the technical field of audio acquisition and noise reduction, and discloses a microphone array method for effectively reducing high-speed wind noise, which realizes the elimination of wind noise and echo and the accurate pickup of human voice in a high-speed motion scene through the specific layout design of a five-microphone array in combination with a multi-channel audio data fusion algorithm. The problems that an existing microphone array cannot adapt to a 40km / h high-speed wind noise scene and the sound receiving definition is insufficient are solved. According to the invention, five microphone arrays are deployed according to functions, the MIC1 is specially used for collecting loudspeaker echoes, the MIC2 and the MIC5 symmetrically capture multi-directional wind noise, the MIC3 and the MIC4 accurately pick up human voice in a leeward area, various signals are collected in a classified manner and do not interfere with each other, and a pure data basis is provided for subsequent noise reduction processing; particularly, the MIC3 and the MIC4 form a straight line with the mouth of a wearer and are positioned in a leeward area shielded by the lenses, so that the wind noise interference is greatly reduced, and the voice acquisition definition is improved.

[Problem] To provide an audio system that outputs, at an appropriate sound level, a masking sound that masks music that is being listened by another user. [Solution] An audio signal processing device is provided with: a crosstalk detection unit (1) that extracts, as a crosstalk signal, a component of a second audio signal output to a second seat speaker from among the outputs of a microphone; a noise detection unit (2) that extracts, as a noise signal, a component from the output of the microphone that does not include a component of the second audio signal and a component of the output sound of the first seat speaker but includes a component of ambient noise; a gain calculation unit (3) that calculates the gain adjustment amount of the obscured noise signal so that the obscured noise signal is larger than the difference between the crosstalk signal and the noise signal by a predetermined degree; a gain adjustment unit (42) that adjusts the gain of the masking noise signal by the calculated gain adjustment amount; and a synthesis unit that synthesizes the gain-adjusted masking noise signal and the first audio signal and outputs the synthesized signal to the first seat speaker.

To provide an earphone which generates at least a bone conduction sound.SOLUTION: An earphone 10 includes a support assembly which is installed as a head beam assembly 12, and a core module 11 which is connected to the support assembly and allows the external auditory canal of the ear of a user to be opened. The support assembly supports the core module to be mounted at a mounting position. The core module includes: a core housing, an energy conversion device, and a vibration panel, where the energy conversion device is disposed in a receiving cavity of the core housing, the vibration panel is connected to the energy conversion device, and configured to transmit mechanical vibration generated by the energy conversion device to the user.SELECTED DRAWING: Figure 12

This paper provides techniques for identifying the user of audio earphones. Specifically, the wearer's head filters an audio signal, and the head's audio filtering capabilities are used as a biometric feature. One earphone can be used as an audio transmitter and the other as an audio receiver. A broadband tone can be generated by the speaker in one earphone and received at the microphone of the other earphone. The received tone is filtered by the user's head, and the head characteristics of the received filtered tone can be used to identify the user. In particular, the material properties of the user's head modify the signal, so that the signal received at the microphone of the other earphone differs from the transmitted signal.The differences are unique to the user's head due to physiological variations between individuals and can be used to identify the user.

A microphone assembly includes a microphone array and a mixer. The microphone array includes first, second, and third microphones. The mixer includes an analog switch assembly and an amplifier assembly configured to mix recording signals of the microphones in different manners to realize different recording patterns of the microphone array. The different recording patterns realizable by the microphone array may include the cardioid pattern, the omnidirectional pattern, the stereo pattern, and the bidirectional pattern.

Techniques are described for wirelessly receiving, at first and second speaker assemblies, signals representing left and right audio channels from each of first and second sources, and using the signals, simultaneously playing, on both speaker assemblies, music, computer game audio, computer chat audio, and speech picked up by speaker microphones.

Methods and apparatuses for capturing and encoding ambisonic audio are described herein. An example ambisonic microphone may comprise multiple sets of microphone capsules arrayed around a central axis. Each set may include two microphone capsules positioned along a first axis that is perpendicular to the central axis and two microphone capsules positioned along a second axis that is perpendicular to the central axis and spaced apart from the first axis by a predetermined distance. The multiple sets may be arranged in a sequence such that each set of the multiple sets is rotated about the central axis in the same angular direction by a predetermined angle between adjacent sets in the sequence.

The invention relates to a method for auditory evaluation of a predefined space (2) comprising the following steps: - taking an acoustic sample over a predefined duration with an acoustic sensor (1), in the predefined space (2), - converting the sample into a data stream corresponding to a stream of signals arriving at the brain with frequency channels, - evaluating for each channel the signal stream according to a biological value scale by determining an acoustic nuisance index for each channel, - selecting a characterization profile of said person including physiological, temporal and psychic characteristics, - selecting a characterization profile of the space including characteristics of an activity present in the space and of a place, - providing the acoustic nuisance indices of the channels as well as the characterization profile of said person and the characterization profile of the space as input to an artificial neural network.Figure 2.

We offer high-quality, stable, discreet, open-back headphones that fit comfortably in the ear. [Solution] The body 14 of the open-type headphones 10 is coupled to an acoustic module 12 and includes a first portion 16 that passes over the outside of the ear and a second portion 18 that is positioned behind the outer ear, the acoustic module 12 is seated in the concha 51 of the outer ear 50 and has a first sound-emitting opening 100 that emits sound generated by the acoustic transducer of the acoustic module 12.

Disclosed are an electronic device and a control method. The electronic device comprises a communication interface for communicating with an external apparatus in a UWB manner, a microphone, a camera, a sensor, a display, and a processor, wherein the processor obtains distance information and angle information with respect to the external apparatus, on the basis of data received from the external apparatus, obtains photographing direction information, on the basis of a detected direction of the camera, identifies the external apparatus, on the basis of the acquired photographing direction information, distance information, and angle information, controls the microphone or communication interface to acquire an audio signal, on the basis of a distance from the external apparatus, and controls the display to display a UI indicating the amplitude of the acquired audio signal, together with a displayed image.

The present disclosure provides AR glasses, an audio enhancing method and device therefor, as well as a readable storage medium. The audio enhancing method for AR glasses worn by a user in a surrounding environment includes: detecting a distribution of sound sources in the surrounding environment using a microphone array; marking a position of each sound source in the distribution of sound sources on lenses of the AR glasses; locking onto one of the distribution of sound sources as target sound source based on an eye gazing direction of the user; extracting and enhancing an audio component associated with a voiceprint characteristic of the target sound source from an audio signal received by the microphone array, configures to obtain an enhanced audio signal; and outputting the enhanced audio signal to the user through an in-ear headphone.

Generating a personalised Head-Related Transfer Function (HRTF) for a user, the system comprising: a sound source at a first position; a user device comprising: a left microphone arranged to be at a l