99results about "Reverberation time" patented technology

The invention provides an equipment fault detection method, a robot and electronic equipment, and the method comprises the steps: obtaining the sound data of target equipment, and carrying out the abnormality prediction of the sound data through a sound abnormality detection model, and obtaining a prediction result; obtaining a visible light image and an infrared image of the target equipment, and determining the real-time temperature of the target equipment according to the visible light image and the infrared image; and performing fault detection on the target equipment according to the prediction result and the real-time temperature of the target equipment. Mutual verification is carried out through combination of sound abnormity of the target equipment and real-time temperature detection determined by an image, fault detection is carried out on the target equipment according to a prediction result of sound data and the real-time temperature of the target equipment, and the multi-modal data of the real-time temperature determined by the sound data and the image can evaluate the health condition of the equipment more comprehensively. The probability of false alarm possibly caused by environmental noise by depending on sound alone is reduced, and the accuracy of fault detection on the equipment is greatly improved.

Examples of the disclosure describe systems and methods for estimating acoustic properties of an environment. In an example method, a first audio signal is received via a microphone of a wearable head device. An envelope of the first audio signal is determined, and a first reverberation time is estimated based on the envelope of the first audio signal. A difference between the first reverberation time and a second reverberation time is determined. A change in the environment is determined based on the difference between the first reverberation time and the second reverberation time. A second audio signal is presented via a speaker of a wearable head device, wherein the second audio signal is based on the second reverberation time.

A method for generating an impulse response representing a sound wave propagation from at least one sound source received at a listening point in a room includes obtaining the generated impulse response at the listening point in the room from a neural network architecture by providing at least the position of the listening point as input. The generated impulse response is generated using a neural network architecture. The network is trained by obtaining a 3D model of the room including the at least one sound source emitting sound in the room and obtaining a training group of simulated impulse responses, wherein each simulated impulse response is generated for a respective predefined listening point in the 3D model of the virtual room. An autoencoder is trained by training an encoder of the autoencoder by using the training group of simulated impulse responses as input in order to obtain a corresponding training group of compressed simulated impulse response as outputs and training a decoder of the autoencoder by using the training group of compressed impulse responses as input in order to obtain a corresponding training group of uncompressed simulated impulse response as outputs. An IR neural network is trained using the training group of compressed simulated impulse responses of the autoencoder and the corresponding position of the predefined listening points as input.

An acoustic processing device includes: an acquisition unit that acquires a recommended environment defined for each content, the recommended environment including an ideal arrangement of speakers in a space in which the content is reproduced; a measurement unit that measures a position of a listener located in the space, the number and arrangement of the speakers, and a space shape; and a correction unit that corrects audio that is observed at the position of the listener, the audio included in the content emitted from a speaker located in the space, to audio to be emitted from a virtual speaker ideally disposed in the recommended environment on the basis of information measured by the measurement unit.

The invention provides an ocean reverberation time domain waveform realization method and system based on convolution operation, and the method comprises the steps: generating a one-dimensional random number sequence meeting a random process, and the data size of the random sequence is the same as the data size of a reverberation time domain waveform to be realized; generating a sound source signal time domain waveform; performing convolution operation on the random number sequence and the sound source signal time domain waveform to obtain a stabilized time domain waveform signal subjected to reverberation intensity stabilization processing; generating an ocean reverberation intensity attenuation curve according to the reverberation intensity theoretical model; and performing intensity weighting on the stabilized time-domain waveform signal subjected to reverberation intensity stabilization processing by using the attenuation curve to obtain a reverberation time-domain waveform. The method has the advantages that cross superposition of scattering echoes of different discrete points does not need to be considered; the calculation speed is high, the requirement for the data cache capacity is small, and the amplitude and the phase of each discrete point echo and the broadening of a transmitted signal do not need to be calculated; the method is suitable for realizing reverberation time domain waveforms in various waveguide environments.

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.

It is proposed to process sound signals acquired by an array of microphones and generated by a sound source to acoustically characterize a space (ESP) comprising the array (MIC) and the sound source and delimited by walls (PAR). A time-frequency transform is applied to the acquired signal and a generalized velocity vector is expressed in the frequency domain based on the acquired signal. Specifically, the generalized velocity vector is represented in the time domain v (t) in the form of a series of peaks comprising at least one peak associated with reflections from the wall and time abscissas associated with the delay TAU1, and the expression of the generalized velocity vector in the time domain is modeled by autoregressive moving average ARMA, the autoregressive moving average (ARMA) is defined by an autoregressive filter (AR) and a moving average (MA). Thus, it is provided that the acquired sound signal is processed by applying an autoregressive filter (AR) to obtain an impulse response characterizing the space (ESP) and obtained from the moving average (MA).

The application relates to a two-step correlation-based chirp ultrasonic signal time-of-flight estimation method, and belongs to the method for measuring the time-of-flight of ultrasonic waves. The coarse correlation is used to obtain the non-accurate TOF of the chirp ultrasonic signal, and the fine correlation is used to obtain the accurate TOF of the chirp ultrasonic signal. The application has the advantages that the accurate time-of-flight TOF is obtained through two-step correlation calculation, the amount of calculation is effectively reduced due to the small amount of data participating in the correlation calculation, and a detection method with low calculation amount is provided for the application fields of distance measurement, multi-target tracking and positioning and other ultrasonic time-of-flight detection-based application fields.

Systems and methods of object classification at point of sale are provided. In one exemplary embodiment, a method is performed by a POS system having processing circuitry operationally coupled to a transducer operable to radiate a pressure wave and a sensor operable to sense a pressure wave. Further, the transducer and the sensor are positioned on or about the POS system so that a field of radiation of the transducer and a field of detection of the sensor are directed towards a certain region of the POS system. The method includes receiving, from the sensor, a reflected signal that represents a pressure wave radiated by the transducer that is reflected from a surface of an object when in or about the certain region of the POS system so that the object can be classified as at least one of a set of object types based on the reflected signal.

Examples of the present disclosure describe systems and methods for estimating acoustic properties of an environment. In the example method, a first audio signal is received via a microphone of a wearable head device. An envelope of the first audio signal is determined, and a first reverberation time is estimated based on the envelope of the first audio signal. A difference between the first reverberation time and a second reverberation time is determined. A change in the environment is determined based on the difference between the first reverberation time and the second reverberation time. A second audio signal is presented via a speaker of the wearable head device, wherein the second audio signal is based on the second reverberation time.

A method including receiving an audio scene including at least one source captured using at least one near field microphone and at least one far field microphone (1410). The method includes determining at least one room-impulse-response (RIR) associated with the audio scene based on the at least one near field microphone and the at least one far field microphone (1420), accessing a predetermined scene geometry corresponding to the audio scene (1430), and identifying best match to the predetermined scene geometry in a scene geometry database (1430). The method also includes performing RIR comparison based on the at least one RIR and at least one geometric RIR associated with the best matching geometry (1450), and rendering a volumetric audio scene based on a result of the RIR comparison (1460).

In various embodiments, a computer-implemented method comprises emitting a reference audio signal using one or more speakers of an audio processing device located in an acoustic environment, receiving, by each microphone in a microphone pair of the audio processing device, a response of the acoustic environment to the reference audio signal, determining an impulse response of the acoustic environment based on the reference audio signal and the received response of the acoustic environment, determining, based on the impulse response, a plurality of candidate wall distance estimates and a plurality of confidence scores, and determining based on the plurality of candidate wall distance estimates and confidence scores, including candidate wall distance estimates and confidence scores from one or more other audio processing devices, a set of wall distance estimates, each wall distance estimate in the set of wall distance estimates being associated with a different wall in a plurality of walls.

In various embodiments, a computer-implemented method includes: emitting a reference audio signal using one or more speakers of an audio processing device located in an acoustic environment; receiving, by each microphone of a microphone pair of the audio processing device, a response of the acoustic environment to the reference audio signal; determining an impulse response of the acoustic environment based on the reference audio signal and the received response of the acoustic environment; determining a plurality of candidate wall distance estimates and a plurality of confidence scores based on the impulse response; and determining a set of wall distance estimates based on a plurality of candidate wall distance estimates and a plurality of confidence scores, including candidate wall distance estimates and confidence scores from one or more other audio processing devices, each wall distance estimate of the set of wall distance estimates is associated with a different wall of the plurality of walls.

A sound production apparatus (100) includes an acquisition unit (131) configured to acquire space information indicating a region of a three-dimensional virtual space including a sound source object and one or more three-dimensional objects having a first characteristic as an acoustic characteristic, and coordinate information indicating a configuration of the three-dimensional object, a display control unit configured to display the three-dimensional virtual space, an input unit (133) configured to input setting of a characteristic related to late reverberation of the three-dimensional virtual space, a change unit (134) configured to change the first characteristic related to at least one of the three-dimensional objects to a second characteristic based on a characteristic related to late reverberation, and an output control unit (135) configured to output in a switchable way a first reproduction sound synthesized based on the characteristic related to late reverberation and the first characteristic and a second reproduction sound synthesized based on the characteristic related to late reverberation and the second characteristic.

The method includes receiving an image of a real-world environment, classifying the image using a machine learning classifier to generate classifications associated with sound presets for simulating an acoustic environment, the sound presets each including acoustic parameters that describe sound reverberation, and selecting a sound preset from the sound presets based on the classification.

A method for processing an audio signal (400) in accordance with a room impulse response is described. The method includes separately processing the audio signal (400) with an early part and a late reverberation of the room impulse response, wherein the separate processing includes processing the audio signal (400) with the early part of the room impulse response during a first (422) process, processing the audio signal (400) with the late reverberation of the room impulse response or with the synthetic reverberation during a second process (424) that is different and separate from the first process (422), and changing from the first process (422) to the second process (424) at a transition from the early part to the late reverberation in the room impulse response. The method further includes combining (432) the audio signal (428) processed with the early part of the room impulse response and the audio signal (430) processed with the late reverberation of the room impulse response or with the synthetic reverberation. The transition from the early part to the late reverberation in the room impulse response is a time when a correlation measure reaches a threshold. The correlation measure describes with regard to the room impulse response a similarity of a decay in acoustic energy comprising an initial state and of the decay in acoustic energy starting at a point in time, said point in time following the initial state over a predefined frequency range. The threshold is set dependent on the correlation measure for said point in time, said point in time being a time of a selected one of the early reflections in the early part (301, 302) of the room impulse response, and the selected one of the early reflections is the first reflection.

In at least one embodiment, an apparatus for generating audio in a listening room of a vehicle is provided. The apparatus includes a memory device and at least one audio controller. The at least one audio controller is programmed to receive an incoming audio signal comprising primary audio data from an audio source for playback, and to receive a first control signal indicative of a desired location for playback of the primary audio data. The at least one audio controller is further programmed to adjust reverberation of at least the incoming audio signal to play the primary audio data in a desired location of the vehicle and to receive a second control signal indicative of a desired natural sound to play in the listening room. The at least one audio controller is further programmed to play the desired natural sound along with primary audio data in the desired venue.

A method for reverberation suppression may involve receiving an input audio signal. The method may involve calculating an initial reverberation suppression gain for the input audio signal for at least one frame of the input audio signal. The method may involve calculating at least one adjusted reverberation suppression gain, where the at least one adjusted reverberation suppression gain adjusts at least one of: 1) a reverberation suppression decay based on a reverberation intensity detected in the input audio signal; 2) gains applied to different frequency bands of the input audio signal based on an amount of room resonance detected in the input audio signal; or 3) a loudness of the input audio signal based on a direct part of the input audio signal. The method may involve generating an output audio signal by applying the at least one adjusted reverberation suppression gain to the input audio signal.

Method for determining a sound field in an enclosed space or at a target position within the enclosed space, the method has the following steps: performing an acoustic measurement at at least one measurement position within the enclosed space or at a single one measurement position within the enclosed space to obtain an interim value set; estimating a surface impedance as first parameter describing the enclosed space based on the interim value set; determining a model of the enclosed space based on the first parameter; and estimating a sound field throughout the enclosed space based on the model of the enclosed space, the sound field describing a sound characteristic for a plurality of positions within the enclosed space.

According to an example aspect of the present invention, there is provided a method of producing a report for optimizing an installation of a loudspeaker system (100) comprising said at least one loudspeaker (101, 102, 103) 5 in a monitoring space. The disclosed method involves providing audio data representative of the performance of the loudspeaker system (100), performing an analysis (1030), wherein a value for at least one audio parameter (1... n) from the audio data is analysed processed into 0 results, and outputting (1039) the compiled results.

A plurality of microphone signals of a microphone array may be obtained. An environment change may be detected based on the microphone signals. In response, a reverberation time environment may be determined. The reverberation may be used to modify a playback audio signal.

In various embodiments, a method for estimating a room layout includes receiving candidate wall distance estimates and a confidence score for walls in an acoustic environment; receiving device position data of the device; determining a location of each wall relative to each audio processing device based on the device location data; generating a room layout matrix including the candidate wall distance estimates, where each row of the room layout matrix is associated with a respective device and each column is associated with a respective wall, and each of the candidate wall distance estimates is associated with the respective wall based on the confidence score; determining a set of wall distance estimates in an ordinal device based on the highest confidence score in each column of the room layout matrix and the location data; and determining a room layout estimate for the acoustic environment based on the set of wall distance estimates.

A method for generating an impulse response representing a sound wave propagation from at least one sound source received at a listening point in a room includes obtaining the generated impulse response at the listening point in the room from a neural network architecture by providing at least the position of the listening point as input. The generated impulse response is generated using a neural network architecture. The network is trained by obtaining a 3D model of the room including the at least one sound source emitting sound in the room and obtaining a training group of simulated impulse responses, wherein each simulated impulse response is generated for a respective predefined listening point in the 3D model of the virtual room. An autoencoder is trained by training an encoder of the autoencoder by using the training group of simulated impulse responses as input in order to obtain a corresponding training group of compressed simulated impulse response as outputs and training a decoder of the autoencoder by using the training group of compressed impulse responses as input in order to obtain a corresponding training group of uncompressed simulated impulse response as outputs. An IR neural network is trained using the training group of compressed simulated impulse responses of the autoencoder and the corresponding position of the predefined listening points as input.

Disclosed is a method for enhancing audio fidelity in a virtual teleconferencing environment (302) comprising recording a first audio signal from a first room (308); performing room impulse response (PIP) analysis for the first room (308) to determine at least one of a direct sound component (406), an early reflections component (408) and a late reverberation component (410); extracting at least the late reverberation component from the determined RIR components; recording a second audio signal from a second room (310); 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 (308) and abstaining from playing in the second room (310). The virtual teleconferencing environment is generated based on the first room (308) to act as an audio source for the second audio signal.