431results about "Microphone spatial/constructional arrangements" patented technology

A method for estimation of one or more audio spatialization parameters for a specific user, a training method and an audio system is provided, wherein the estimation method comprises obtaining audio data comprising first second audio data by obtaining the first audio data from a first microphone arranged near, in, or at a first ear canal of the target user and the second audio data from a second microphone arranged near, in, or at a second ear canal of the target user; and providing the one or more audio spatialization parameters comprising: applying a model to the audio data for provision of a parameter estimate of the one or more audio spatialization parameters; determining the one or more audio spatialization parameters based on the parameter estimate; and outputting the one or more audio spatialization parameters.

A method, computer program product, and computing system for generating a plurality of acoustic relative transfer functions associated with a plurality of audio acquisition devices of an audio recording system deployed in an acoustic environment. Acoustic relative transfer functions of at least a pair of audio acquisition devices of the plurality of audio acquisition devices may be compared. Location information associated with an acoustic source within the acoustic environment may be determined based upon, at least in part, the comparison of the acoustic relative transfer functions of the at least a pair of audio acquisition devices of the plurality of audio acquisition devices.

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.

Systems and techniques are provided for performing spatial audio recording. For instance, a process can include detecting an occlusion for at least one audio frame of one or more audio frames associated with a spatial audio recording. During the spatial audio recording, the process can further include selecting, based on detection of the occlusion, at least one of an occluded spatial filter for the one or more audio frames or a non-occluded spatial filter for the one or more audio frames.

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 presents a communication device configured to removably attach to a mask, and to receive, transmit, and amplify sound and / or speech from a wearer.

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.

A method may include capturing an image at a camera included at an information handling system, the camera coupled to a vision system. A position of a user relative to a display device may be determined based on analysis of the image by the vision system. The method may further include adjusting properties of an audio signal provided to a speaker based on the position of the user.

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.

The present invention relates to a method for obtaining a position of a sound source relative to a dedicated reference point, where a first audio signal and a number of second audio signals are recorded synchronously in time. The position can be obtained by applying an estimated filter to a correlation signal derived by correlation of the first audio signal with at least one of the number of second audio signals in the frequency domain. Two timing values ​​are derived for at least one filtered correlation signal that exceeds a dedicated threshold in the time domain. Then, a distance between the dedicated reference point and the sound source is based on each obtained first and second timing value.

A head-mounted device for generating binaural audio content includes a first rod coupled to a first microphone housing enclosing a first front microphone and a first rear microphone, respectively, that generate acoustic signals. The first microphone housing includes a first front port facing downward and a first rear port facing rearward. The apparatus includes a second rod coupled to a second microphone housing enclosing a second front microphone and a second rear microphone, respectively, that generate acoustic signals. The second microphone housing includes a second front port facing downward and a second rear port facing rearward. In one embodiment, an apparatus includes a binaural audio processor including a beamformer and a storage device. The beamformer generates a first beamformer signal based on acoustic signals from the first front microphone and the first rear microphone and a second beamformer signal based on acoustic signals from the second front microphone and the second rear microphone. A storage device stores the first and second beamformer signals as a two-channel file.

To correct displacement between a photographing direction and a sound collecting direction.SOLUTION: An imaging apparatus has: connection means that connects to a sound collecting device that collects voice data in different directions; voice processing means that generates three-dimensional acoustic data on the basis of the voice data acquired from the sound collecting device; and correction means that corrects the three-dimensional acoustic data on the basis of the difference between the photographing direction of the imaging apparatus and a sound collecting direction of the sound collecting device.SELECTED DRAWING: Figure 1

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.

Disclosed is a method for enhancing audio fidelity in a virtual teleconferencing environment including recording a first audio signal from a first room; performing room impulse response (RIR) analysis for the first room to determine at least one of a direct sound component, an early reflections component and a late reverberation component; extracting at least the late reverberation component from the determined RIR components; recording a second audio signal from a second room; de-reverberating the second audio signal and convolving the de-reverberated second audio signal with at least the late reverberation component to form a third audio signal. Playing the third audio signal in the first room and abstaining from playing in the second room. The virtual teleconferencing environment is generated based on the first room to act as an audio source for the second audio signal.

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.

An audio processor for providing a plurality of loudspeaker signals, or loudspeaker feeds, on the basis of a plurality of input signals, like channeled signals and / or object signals, is configured to obtain information about the position of a listener, to obtain an information about the position of a plurality of loudspeakers, or sound transducers, which may be placed within the same containment, e.g. a soundbar, to dynamically adapt an allocation of objects and / or channel objects and / or adapted signals, like adapted channel signals, derived from the input signals, like channel signals or channel objects, or like upmixed or downmixed signals, to loudspeakers, and to render the objects and / or the channel objects and / or the adapted signals derived from the input signals, in dependence on the information about the positions of the listener and of the loudspeakers, and on the allocation, in order to obtain the loudspeaker signals, such that a rendered sound follows a listener.

A head-mounted device for generating binaural audio content includes a first rod coupled to a first microphone housing enclosing a first front microphone and a first rear microphone, respectively, that generate acoustic signals. The first microphone housing includes a first front port facing downward and a first rear port facing rearward. The apparatus includes a second rod coupled to a second microphone housing enclosing a second front microphone and a second rear microphone, respectively, that generate acoustic signals. The second microphone housing includes a second front port facing downward and a second rear port facing rearward. In one embodiment, an apparatus includes a binaural audio processor including a beamformer and a storage device. The beamformer generates a first beamformer signal based on acoustic signals from the first front microphone and the first rear microphone and a second beamformer signal based on acoustic signals from the second front microphone and the second rear microphone. A storage device stores the first and second beamformer signals as a two-channel file.

An audio generation system is provided to enable coordinated control of multiple IoT devices for audio collection and distribution of one or more audio sources according to location and user preference. The audio generation system enables a location sensitive acoustic control of sound, both as a shaped envelope for a particular source, and as an individualized experience. The audio generation system also facilitates an interactive visual system for visualization and manipulation of the audio environment including via the use of augmented reality and / or virtual reality to depict soundscapes. The audio generation system can also facilitate a system for improving and achieving an audio environment (sound influence zone) and an intuitive way to understand where sounds will be heard.

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 invention relates to spatial audio parameters and associated spatial audio playback. An apparatus (105) includes at least one processor and at least one memory including computer program code, the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus (105) to at least: for two or more microphone audio signals (102), transmit the two or more microphone audio signals (102); determining at least one spatial audio parameter for providing spatial audio reproduction (108, 304); based on the two or more microphone audio signals (102), at least one coherence parameter (112, 114) associated with a sound field is determined such that another sound field is configured to be reproduced based on the at least one spatial audio parameter (108, 304) and the at least one coherence parameter (112, 114).

The present disclosure provides an earphone including a sound production component, an ear hook, and a microphone assembly. The microphone assembly includes a first microphone and a second microphone. The sound production component or ear hook includes a first sound hole and a second sound hole corresponding to the first microphone and second microphone, respectively. An extension line of a line connecting a projection of the first sound hole on a sagittal plane of the user and a projection of the second sound hole on the sagittal plane has an intersection point with a projection of an antihelix, a ratio of a first distance between the projection of the first sound hole on the sagittal plane and the projection of the second acoustic hole on the sagittal plane to a second distance between the projection of the second acoustic hole on the sagittal plane and the intersection point is 1.8-4.4.

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.

An electronic device according to various embodiments disclosed in the present document may comprise a communication circuit, a memory, and a processor operatively connected to the communication circuit and the memory. The memory can store instructions that, when executed by the processor, enable the electronic device to: receive, through the communication circuit, two or more pieces of audio data simultaneously obtained by one or more external electronic devices; use user information acquired from the memory so as to identify user personal HRTF information acquired on the basis of reference HRTF information; identify sound source generation location information of the two or more pieces of audio data on the basis of the two or more pieces of audio data; perform inverse filtering on the two or more pieces of audio data on the basis of the user HRFT information so as to generate an audio signal; and transmit the audio signal and the sound source generation location information to another electronic device through the communication circuit.

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.