292results about "Microphone spatial/constructional arrangements" patented technology

The invention provides electronic equipment and a pickup device, the electronic equipment is provided with an appearance surface, the appearance surface is provided with a first pickup port and a plurality of second pickup ports distributed at intervals, and each second pickup port and the first pickup port are arranged at intervals; a microphone, a first sound channel and a plurality of second sound channels are arranged in the electronic equipment, the first sound channel is communicated with the first pickup port and a first cavity of the microphone, the second sound channels are communicated with the second pickup ports in a one-to-one correspondence mode, and the second sound channels are communicated with a second cavity of the microphone. And the first cavity of the microphone and the second cavity of the microphone are respectively positioned on two sides of a vibrating diaphragm of the microphone. A plurality of pickup vectors can be formed between the plurality of second pickup ports and the first pickup port. A vector obtained by adding the plurality of pickup vectors is consistent with the sensitivity direction of the microphone. The microphone of the electronic equipment can better receive sound from the sensitivity direction of the microphone, and the microphone can realize directional pickup.

Conferencing systems and methods configured to generate true talker coordinates for use in camera tracking of talkers and objects in an environment and other room intelligence use cases are disclosed. The initial configuration and ongoing usage of conferencing systems can be improved by detecting and converting the locations of objects and talkers in an environment into a common coordinate system. The amount of time and effort by installers, integrators, and users, can be reduced leading to increased satisfaction with installation and usage of the conferencing system.

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.

The invention relates to a method for obtaining a position of a sound source relative to a dedicated reference point. A first and a plurality of second sound signals are recorded which are synchronized in time. The position can be obtained by applying an estimated filter to a correlated signal derived by correlation of the first sound signal with at least one of the plurality of second sound signals in the frequency domain. Two timing values are derived in the at least one filtered and correlated signal exceeding a dedicated threshold in the time domain. Then the distance between the dedicated reference point and the sound source based on the respective obtained first timing value and second timing value.

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.

Examples of the disclosure relate to a method, apparatus and computer program, the method comprising:obtaining audio signals wherein the audio signals represent spatial sound and can be used to render spatial audio using linear methods;obtaining spatial metadata corresponding to the spatial sound represented by the audio signals; and associating the spatial metadata with the obtained audio signals so that in a first rendering context the obtained audio signals can be rendered without using the spatial metadata and in a second rendering context the obtained audio signals can be rendered with using the spatial metadata.

Conferencing systems and methods configured to generate true talker coordinates for use in camera tracking of talkers and objects in an environment and other room intelligence use cases are disclosed. The initial configuration and ongoing usage of conferencing systems can be improved by detecting and converting the locations of objects and talkers in an environment into a common coordinate system. The amount of time and effort by installers, integrators, and users, can be reduced leading to increased satisfaction with installation and usage of the conferencing system.

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).

A computer system and a processing method thereof of sound signal are provided. The computer system includes a platform path controller (PCH), a high-definition audio (HDA) codec, and a digital microphone. The HDA codec is coupled to the PCH. The digital microphone is coupled to the PCH and the HDA codec. The digital microphone is used to generate sound signal. The HDA codec processes the sound signal from the digital microphone. Accordingly, the audio recording with high quality could be provided.

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

A method, computer program product, and computing system for encoding audio encounter information of a reference audio acquisition device of a plurality of audio acquisition devices of an audio recording system, thus defining encoded reference audio encounter information. Location information may be estimated, via a machine vision system, for an acoustic source within an acoustic environment. One or more acoustic relative transfer functions may be selected from a plurality of acoustic relative transfer functions for the plurality of audio acquisition devices of the audio recording system based upon, at least in part, the location information. The encoded reference audio encounter information and a representation of the selected one or more acoustic relative transfer function may be transmitted.

A terminal apparatus includes a first structural component, a second structural component, a third structural component, and at least one acoustic element, where the second structural component is located between the first structural component and the third structural component, and the first structural component, the second structural component, and the third structural component surround and form a cavity; the acoustic element is accommodated in the cavity; at least one sound emission slot is formed between the first structural component and the second structural component and / or between the second structural component and the third structural component, a sound guide slot is in communication with the sound emission slot, the sound emission slot is located on a side of the sound guide slot that faces away from the acoustic element, and the acoustic element transmits sound to the outside through the sound guide slot and the sound emission slot.

A display apparatus(10) includes a first display device(12) having two first sound reception units(20), a second display device(14) having two second sound reception units(22), and a sound reception processor(18). The second display device(14) is connected to a side of the first display device(12). The sound reception processor(18) is coupled to the first sound reception units(20) and the second sound reception units(22). The sound reception processor(18) turns on at least one of the first sound reception units(20) to receive sound in a first normal direction of the first display device(12) for generating a first directional sound signal, turns on at least one of the second sound reception units(22) to receive sound in a second normal direction of the second display device(14) for generating a second directional sound signal, and generates a stereo signal according to the first directional sound signal and the second directional sound signal.

A signal processing system including: a measurement device configured to acquire measurement data regarding a transfer characteristic; a first acquisition device configured to acquire a first audio signal; and a control device configured to be connected to each of the measurement device and the first acquisition device through a network, wherein the control device corrects the first audio signal on a basis of the measurement data acquired by the measurement device and generates a second audio signal.

The present invention relates to a method comprising calculating a set of head-related transfer function (HRTFs) based at least in part on audio data of test sounds captured at different orientations of a headset worn by a user, the set of HRTFs individualized to the user while wearing the headset. The method further includes determining a portion of the set of HRTFs to discard using a distortion mapping that identifies one or more distortion regions that are commonly distorted by the headset having a form factor that occludes at least a portion of a face of the user. The portion of the set of HRTFs is then discarded to create an intermediate set of HRTFs, the discarded portion corresponding to the one or more distortion regions. Then, one or more HRTFs are generated that correspond to the discarded portion using at least some of the intermediate set of HRTFs. At least some of the intermediate set of HRTFs are combined with the generated one or more HRTFs to create a customized set of HRTFs that are individualized to the user and mimic actual HRTFs of the user without a headset, and the customized set of HRTFs is provided to the headset.

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.