73 results about "Acoustic space" patented technology

The embodiment of the invention provides an audio processing method and device, electronic equipment and a storage medium. The method comprises the following steps: determining a target pulse response of a first audio signal, wherein the target pulse response is a stereo pulse response used for simulating sound reflection and attenuation effects of the first audio signal in an acoustic space; performing convolution processing on the first audio signal and the target pulse response to obtain a second audio signal; and performing audio mixing on the first audio signal and the second audio signal to output a third audio signal. According to the scheme provided by the invention, the stereo pulse response data capable of simulating the reflection and attenuation effects of the sound in the space can be synchronously determined when reverberation is carried out on the audio signal, and reverberation is carried out on the audio signal by using the stereo pulse response data, so that the sound is heard to be generated in different space environments; the reverberation effect of the audio signal is more vivid, the sense of space and the sense of layering of the sound are increased, and the sound is fuller and stereoscopic.

A loudspeaker system and method can measure a specific user's hearing and implement compensatory or corrective processing to address the user's hearing deficiencies. The user's hearing acuity may be determined by (i) headphone techniques in which frequency-based hearing thresholds are determined; (ii) a loudspeaker system set up as intended for use in an acoustic space to emit tonal stimuli; or (iii) inducing, acquiring, and interpreting otoacoustic emissions. Once the hearing profile is determined, appropriate signal processing parameters can be used to generate a corrected audio output signal adjustment. The administered hearing test results can include a plurality of hearing loudness threshold levels at respective test frequencies. The hearing correction system's corrected audio output can be generated from corrected loudness values at selected correction frequency sub-bands.

The invention relates to a vehicle health dynamic monitoring method and system based on acoustic fingerprints, and belongs to the field of vehicle safety. The method comprises the following steps: capturing voiceprint transmission delay inequality and sensing baseline spacing, calculating a sound source azimuth angle, and positioning an abnormal sound source coordinate; triggering an active voiceprint acquisition protocol based on the abnormal sound source coordinate, acquiring an original abnormal sound signal, and performing environmental noise suppression to obtain a pure signal with a high signal-to-noise ratio; starting an acoustic fingerprint hierarchical scanning protocol, decomposing the high-signal-to-noise-ratio pure signal into fine sub-bands, extracting a key monitoring frequency band, and obtaining a frequency band attention index based on the key monitoring frequency band; and when the frequency band attention index exceeds a preset threshold range, outputting a troubleshooting instruction and generating a reverse sound wave signal and a vibration hedging amplitude. According to the invention, high-precision acoustic space perception and adaptive noise suppression optimization are realized, and dynamic monitoring of vehicle health is effectively carried out.

The invention provides a control method and a control system for building an independent acoustic space of an automobile cabin, which are applied to the technical field of sound field control of the automobile cabin, and are used for inputting a multi-channel vibration signal acquired in real time into a deep learning noise prediction model to obtain an in-automobile background noise prediction value at a future moment; the prediction noise is filtered by the secondary path estimation model and then is used as a feedforward reference signal for active noise reduction, and the prediction noise and a residual signal collected by a bright area error microphone are jointly used for weight updating of an adaptive filter; the updated filter generates a reverse-phase noise elimination signal and the reverse-phase noise elimination signal is played by a secondary loudspeaker in real time so as to counteract background noise of the bright area; and closed-loop adjustment of the noise reduction effect is formed based on the sound pressure level contrast ratio of the bright and dark areas, so that the independent acoustic space is kept stable in the time-varying environment of vehicle driving. According to the method, the low noise of the bright area and the isolated sound field of the dark area in the vehicle are realized by fusing deep learning noise prediction and active noise control, and the personalized auditory experience in the vehicle can be remarkably improved.

A display method is performed to easily understand a placement of audio equipment for realizing a target sound pressure distribution in an acoustic space. The display method includes receiving an acoustic space and a target sound pressure distribution in the acoustic space. The display method further includes calculating, based on a prescribed model, a placement distribution of speakers or microphones, which corresponds to the target sound pressure distribution, in the acoustic space. The display method further includes displaying the calculated placement distribution.

A loudspeaker system and method can measure a specific user's hearing and implement compensatory or corrective processing to address the user's hearing deficiencies. The user's hearing acuity may be determined by (i) headphone techniques in which frequency-based hearing thresholds are determined; (ii) a loudspeaker system set up as intended for use in an acoustic space to emit tonal stimuli; or (iii) inducing, acquiring, and interpreting otoacoustic emissions. Once the hearing profile is determined, appropriate signal processing parameters can be used to generate a corrected audio output signal adjustment. The administered hearing test results can include a plurality of hearing loudness threshold levels at respective test frequencies. The hearing correction system's corrected audio output can be generated from corrected loudness values at selected correction frequency sub-bands.

A spatial audio processing system operable to enable audio signals to be spatially extracted from, or transmitted to, discrete locations within an acoustic space. Embodiments of the present disclosure enable an array of transducers being installed in an acoustic space to combine their signals via inverting physical and environmental models that are measured, learned, tracked, calculated, or estimated. The models may be combined with a whitening filter to establish a cooperative or non-cooperative information-bearing channel between the array and one or more discrete, targeted physical locations in the acoustic space by applying the inverted models with whitening filter to the received or transmitted acoustical signals. The spatial audio processing system may utilize a model of the combination of direct and indirect reflections in the acoustic space to receive or transmit acoustic information, regardless of ambient noise levels, reverberation, and positioning of physical interferers.

A microphone transducer is provided, the microphone transducer comprising a housing and a transducer assembly supported within the housing and defining an internal acoustic space. The transducer assembly includes a magnet assembly, a diaphragm disposed adjacent the magnet assembly and having a front surface and a rear surface, and a coil attached to the rear surface of the diaphragm and capable of moving relative to the magnet assembly in response to acoustic waves impinging on the front surface. The transducer assembly further includes a primary port establishing acoustic communication between the internal acoustic space and an external cavity at least partially within the housing, and a secondary port located at the front surface of the diaphragm.

A method for self-calibrating a sound pickup process that uses a microphone array in a wearable device that also includes a loudspeaker, where the microphone array being in a physical arrangement with respect to the loudspeaker. The method obtains, for each of several microphones of the microphone array, one or more transfer functions that each represent a response of the microphone to sound from a position in an acoustic space. The method determines whether a physical arrangement of the microphone array with respect to the loudspeaker has changed and adjusts the transfer function, for at least one of the microphones of the several microphones, in response to determining that the current physical arrangement of the microphone array with respect to the loudspeaker has changed.

The present application belongs to the technical field of audio processing, and particularly relates to an audio reproduction method and system for live sound field data collection and digital signal processing. The method comprises collecting sound field data of a real acoustic space and generating a dedicated sound field parameter model; receiving an original digital audio signal and a user instruction, obtaining a dedicated sound field parameter model corresponding to the user instruction as a target dedicated sound field parameter model; performing convolution processing on the original digital audio signal and the target dedicated sound field parameter model to obtain a processed audio signal with enhanced spatial sense; and outputting the processed audio signal after multi-channel digital amplification, thereby realizing audio reproduction of the acoustic response characteristics of the dedicated sound field. The present application generates a dedicated sound field parameter model based on impulse response data collected in a real acoustic space, and the reproduced reverberation characteristics, spatial sense and sound image positioning are highly consistent with the target space, and the sound field restoration accuracy is significantly improved.

To realize localization of a sound image even with respect to an acoustic component in a low frequency band handled by a subwoofer.SOLUTION: A signal processing device includes a sound source assigning unit 61 that assigns each of a plurality of sound sources corresponding to different positions to one or more subwoofers according to a positional relationship with the sound source among a plurality of subwoofers installed in an acoustic space, and a signal processing unit 62 that generates a plurality of first sound signals respectively corresponding to the plurality of subwoofers from a plurality of sound source signals respectively corresponding to the plurality of sound sources according to a result of the assignment by the sound source assigning unit 61.SELECTED DRAWING: Figure 5

According to one embodiment of the present disclosure, an electronic device can be provided. The electronic device may include: a housing including a first speaker hole; and a speaker assembly disposed in the housing and emitting sound to the outside of the electronic device through the first speaker hole. The speaker assembly includes: a speaker including a diaphragm; a first speaker cover facing the diaphragm and spaced apart from the diaphragm; an acoustic space formed between the diaphragm and the speaker cover; a first support structure including a first portion surrounding a portion of the acoustic space; and a step structure disposed adjacent to a portion of the first portion of the first support structure and an area where the first speaker cover contacts each other and spaced apart from the speaker module.

An audio system includes a plurality of subwoofers installed in an acoustic space and a signal processing device. The signal processing device includes one or more memories storing instructions and one or more processors. The one or more processors are configured to assign each of a plurality of sound sources that correspond to different positions, to one or more subwoofers installed in an acoustic space. Each of the plurality of sound sources is assigned based on a positional relation with each of the plurality of sound sources and a plurality of subwoofers installed in the acoustic space and including the one or more subwoofers. The one or more processors are also configured to generate, from a plurality of sound source signals corresponding to respective ones of the plurality of sound sources, a plurality of first sound signals corresponding to respective ones of the plurality of subwoofers.

The invention provides a DAS water content identification method based on shaft axial acoustic attenuation characteristics. The method comprises the following steps: acquiring DAS acoustic time-domain signals of a plurality of axial positions of a shaft by using a distributed optical fiber; performing preprocessing and frequency domain transformation on the time domain signal to obtain an acoustic frequency spectrum distributed along the axial direction of the shaft; according to the law that the acoustic spectrum amplitude changes along with the position, calculating an acoustic space attenuation feature alpha (f) by adopting least square fitting based on the distance; constructing an attenuation spectrum based on the spatial attenuation characteristics, and extracting attenuation characteristic parameters related to the moisture content from the attenuation spectrum; constructing a training set and establishing a support vector machine (SVM) moisture content identification model based on a radial basis function (RBF); and inputting the actual attenuation characteristic parameters into the trained SVM model to obtain the predicted moisture content. The invention further discloses a water content identification system for executing the method and a storage medium, and the system and the storage medium are suitable for identifying the water content of the oil well shaft fluid.

A sound processing method includes obtaining an image of an acoustic space, setting a plane in the acoustic space using the image, setting a virtual speaker in the image of the acoustic space, calculating sound pressure distribution to the plane using characteristics of the virtual speaker, and overlapping the sound pressure distribution and the image of the plane.

A personalized sound management system for an acoustic space includes at least one transducer, a data communication system, one or more processors operatively coupled to the data communication system and the at least one transducer, and a medium coupled to the one or more processors. The processors access a database of sonic signatures and display a plurality of personalized sound management applications that perform at least one or more tasks among identifying a sonic signature, calculating a sound pressure level, storing metadata related to a sonic signature, monitoring sound pressure level dosage levels, switching to an ear canal microphone in a noisy environment, recording a user's voice, storing the user's voice in a memory of an earpiece device, or storing the user's voice in a memory of a server system, or converting received text received in texts or emails to voice using text to speech conversion. Other embodiments are disclosed.

Provided is, for example, an acoustic reproduction device having a structure capable of effectively vibrating particles possessed by the acoustic reproduction device. The acoustic reproduction device includes: a housing having an internal space; a driver unit accommodated in the inner space of the housing; and a powder accommodating portion in which powder is accommodated, the driver unit having a rear surface located on an opposite side of a radiation direction of sound reproduced from the driver unit, and at least a portion of the rear surface being acoustically spatially connected to the powder accommodating portion.

An audio reproduction device (100) according to the present disclosure includes a reception unit (132) that receives, from a user, a request for reproducing a second audio signal that is an audio signal different from a first audio signal that is an original audio signal of content, and a reproduction unit (133) that localizes the second audio signal at an arbitrary position in an acoustic space including an azimuth direction and a height direction and outputs the first audio signal and the second audio signal in parallel when the reception unit receives the request.

This sound signal processing method includes: accepting acoustic space information, first position information indicating a listening position in an acoustic space, and second position information indicating the positions of a plurality of ambient sounds in the acoustic space; accepting a first sound signal of a content sound and second sound signals related to the plurality of ambient sounds; classifying each of the plurality of ambient sounds into one of a plurality of groups on the basis of the first position information and the second position information; subjecting each of the second sound signals classified into the plurality of groups to sound processing for each classified group; and outputting the first sound signal and the second sound signal after the sound processing.

The invention discloses a coffee machine voice interaction method and system, and relates to the technical field of voice interaction. The method comprises the following steps: when the coffee machine does not perform voice interaction, detecting a kitchen space and monitoring environment background sound, and constructing and dynamically updating a kitchen environment acoustic space model at least representing a noise source physical position, acoustic characteristics and a sound reflection path; when a user voice instruction is detected, accurately positioning the mouth position of the user based on a sound reflection path in the model; performing spatial enhancement, active noise cancellation and active echo cancellation on the mixed audio signal containing the user voice instruction in parallel so as to separate out a pure user voice signal; and identifying the pure user voice signal and executing a corresponding coffee making instruction. According to the method, the acoustic space model is constructed and processed in parallel, so that the noise and echo interference problem in a complex acoustic environment of a kitchen is effectively solved, and the voice recognition accuracy and interaction fluency are remarkably improved.

This provides a sound processing method that makes your own performance sound as if it were coming from a predetermined location within the space. [Solution] In the sound signal processing method, the sound signal processing device, which is a general-purpose information processing device, receives acoustic space information, first position information of a first performer in the acoustic space, and second position information of a second performer in the acoustic space, receives a first sound signal related to the performance of the first performer from an audio interface, receives a second sound signal related to the performance of the second performer via a network, applies first binaural processing to the first sound signal based on the first position information, applies second binaural processing to the second sound signal based on the second position information, and outputs the first binaural signal after the first binaural processing and the second binaural signal after the second binaural processing from the audio interface.

The design in question is the interior of a music hall. The area indicated by the solid lines is a series of reflectors on the wall shown on the back side of the drawing, which are installed to create an acoustic space in which sound is evenly distributed across the audience seats.

A personalized sound management system for an acoustic space includes at least one transducer, a data communication system, one or more processors operatively coupled to the data communication system and the at least one transducer, and a medium coupled to the one or more processors. The processors access a database of sonic signatures and display a plurality of personalized sound management applications that perform at least one or more tasks among identifying a sonic signature, calculating a sound pressure level, storing metadata related to a sonic signature, monitoring sound pressure level dosage levels, switching to an ear canal microphone in a noisy environment, recording a user's voice, storing the user's voice in a memory of an earpiece device, or storing the user's voice in a memory of a server system, or converting received text received in texts or emails to voice using text to speech conversion. Other embodiments are disclosed.

The system effectively detects the flow of sound radiated within an acoustic space. [Solution] The computing device (1) includes a sound detection means (110) that detects sound from a sound source (300) at at least two locations (210, 220), and a computing means (120) that performs calculations related to the propagation direction of sound from the sound source based on the times (T0, T1) at which sound is detected at each of the at least two locations. With such a computing device, it becomes possible to intuitively and simply know the flow of sound radiated into the acoustic space.

A sound signal processing method and a sound signal processing apparatus are capable of clearly reproducing sound image localization in a virtual space. A sound signal processing method classifies virtual sound sources representing reflected sound in a target acoustic space into a first virtual sound source that is located between the position of a speaker and the position of a sound reception point and that represents the reflected sound in the target acoustic space, and a second virtual sound source that is located outside the speaker and that represents the first sound source of the reflected sound in the target acoustic space and that represents the second sound source of the reflected sound in the target acoustic space between the position of the speaker and the position of the sound reception point. And a second sound source indicating the reflected sound of the target acoustic space, the position of the first virtual sound source is moved to a position at which playback can be performed by the position of the speaker in the vicinity of the first virtual sound source only in the case of the first virtual sound source.

The invention is suitable for the technical field of intelligent medical treatment, and provides a medical intelligent voice interaction processing system. The method has multiple innovations and advantages. On the aspect of technology combination, acoustic role separation and medical knowledge graph dynamic reasoning are deeply fused for the first time, through the voiceprint biological lock and the UWB positioning technology, master control role sound of a medical scene is accurately locked, and the irrelevant voice mixing rate is greatly reduced through the acoustics and space two-dimensional verification mechanism. In the aspect of algorithm design, a symptom-disease-examination three-dimensional map is constructed by a dynamic necessary examination item prediction algorithm, and a real-time weight calculation model is introduced, so that the limitation of a static rule is broken through, and the comprehensiveness and accuracy of a diagnosis process are improved. In a practical level, a federal learning framework is adopted to realize localized data processing and encryption, a'non-sensitization 'privacy protection mechanism is constructed by combining a'sight locking' missed diagnosis prompt of AR glasses, the information prompt efficiency and the psychological comfort of a patient are considered, and the medical diagnosis and treatment efficiency and accuracy are remarkably improved.

A music synthesizer generates an audio signal using an assembly comprising hundreds or thousands of resonators. These resonators may be based on the analysis of any acoustic space, such as acoustic musical instruments, rooms, recording studios, or music halls. A machine learning network is trained to learn features of the music sound. The feature may be whether the sound is melodious. The network produces an audio effect applied to a selected frequency in the spectrum. An input signal or an excitation signal is provided to the network that processes the input by a trained model of the target audio source and configures the resonator components to generate an output audio signal based on the input signal. The network may be expanded to create novel impulse responses to create tones and timbres unique to existing audio sources, which input signals may include musical tones or include human voice inputs.

Vehicle (1), comprising a vehicle cabin (2) defining an acoustic space having specific aurally perceivable properties in a reference state, comprising an apparatus (3) for outputting an audio signal (4) in the vehicle cabin (2), characterized in that the apparatus (3) for outputting an audio signal (4) in the vehicle cabin (2) comprises:an audio outputting device (6) configured to output an audio signal (4) in the vehicle cabin (2);an acoustic impression modifying device (8) configured to generate an acoustic space modification signal (9) allowing for modifying the aurally perceivable properties of the acoustic space defined by the vehicle cabin (2) with regard to a reference state.

A method for an apparatus for spatial rendering of reverberation, the method comprising: obtaining a bitstream for a virtual scene, the bitstream for a virtual scene comprising at least one reverberation parameter associated with a first acoustic environment, and at least one acoustic space connection information associated with the first acoustic environment; obtaining at least one second acoustic environment and associated position data; associating the second acoustic environment with the first acoustic environment based on at least one of the associated position data and the at least one acoustic space connection information; determining at least first reverberation processing based on the association between the second acoustic environment and the first acoustic environment; and generating a reverberated audio signal based on the at least one reverberation parameter and the at least first reverberation processing applied to at least one sound source audio signal.

An audio signal processing method includes dividing a virtual sound source representing a reflected sound of a target acoustic space into (i) a first virtual sound source located between a position of a speaker and a position of a sound receiving point, the first virtual sound source representing a first sound source of the reflected sound of the target acoustic space, and (ii) a second virtual sound source located such that the speaker is disposed between the second virtual sound source and the sound receiving point, the second virtual sound source representing a second sound source of the reflected sound of the target acoustic space, and, only in a case of the first virtual sound source, moving a position of the first virtual sound source to a reproducible position based on a position of the speaker.