64results about "Reverberation time" patented technology

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).

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.

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.

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.

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 comprising said at least one loudspeaker in a monitoring space. The disclosed method involves providing audio data representative of the performance of the loudspeaker system, performing an analysis, wherein a value for at least one audio parameter from the audio data is analysed processed into results, and outputting the compiled results.

The present invention discloses a method for automatically setting an acoustic signal of a space monitoring device using an acoustic signal, which is a technology for automatically setting the acoustic signal emitted from the space monitoring device so as to more accurately grasp changes in the physical conditions of the monitored space.

Disclosed implementations for determining a reverb characteristic of an environment based on an impulsive sound generated within the environment. The impulsive sound is captured by an audio sensor. The impulsive sound is converted to a scale plotting a decay over time of a measure of sound waves of the impulsive sound. A diffused portion of the impulsive sound that includes the sound waves reflected from the environment is determined. The reverb characteristic of the environment is determined based on the magnitude of the diffused portion.

According to an exemplary aspect of the present invention, a method for creating a report for optimizing the installation of a speaker system (100) having at least one speaker (101, 102, 103) in a monitoring space is provided. The method of the present disclosure includes providing audio data representing the performance of the speaker system (100), performing an analysis (1030) of analyzing and processing values of at least one audio parameter (1 to n) from the audio data to obtain a result, and outputting (1039) the collected result.

It is proposed to process sound signals acquired by an array of microphones and generated by a sound source, with a view to acoustically characterizing a space (ESP) comprising the array (MIC) and the source and bounded by a wall (PAR). A time-frequency transform is applied to the acquired signals, and a generalized velocity vector is expressed in the frequency domain on the basis of the acquired signals. In particular, the generalized velocity vector is expressed in the time domain v(t) in the form of a succession of peaks comprising at least one peak associated with reflection from the wall and with a time abscissa dependent on the delay TAU1, and the expression in the time domain of the generalized velocity vector is modelled by an autoregressive moving average ARMA defined by an autoregressive filter AR and a moving average MA. Thus, provision is made to process the acquired sound signals by applying the autoregressive filter AR, to obtain an impulse response characterizing the space (ESP) and obtained from the moving average MA.

Methods and systems are provided for probabilistically determining an expected field response by a reverberant system that accounts for input conductance frequency uncertainty. One method involves determining a marginal probability density function for input conductance frequency uncertainty of a reverberant system based at least in part on physical dimensions of an enclosure structure, a loss factor and a statistical mean of input excitation energy, determining an unconditional probability density function for the field response of the reverberant system to the excitation energy based at least in part on the input conductance frequency uncertainty, determining an expected field response by the reverberant system with respect to frequency in response to the excitation energy for an input probability value using the unconditional probability density function, and providing a graphical user interface (GUI) display including a graphical representation of the expected field response with respect to frequency.

An apparatus detects complex time-variant electromagnetic disturbances resulting from a high-altitude nuclear electromagnetic pulse (EMP) or solar storm. The device relies on a circuit (104, 1301) to monitor the input power lines (301, 401, 501) for sustained conducted electrical disturbances associated with the E3 phase of an EMP or solar storm. A separate circuit (105, 1302) monitors the power lines and the ambient environment (503) for transient electromagnetic pulse disturbances associated with the E1 and E2 phases of an EMP. A housing (901, 902) has a pair of AC mains wall plug prongs (905) extending from a side. When sustained electrical disturbances or transient electromagnetic pulse disturbances are detected, the apparatus provides a visual alarm (603, 609), audible alarm (608), and discrete indication signal (607) that can be used to disconnect (702) or redirect the flow of electrical power.

To appropriately identify a resonant frequency when a broadcast sound is broadcasted, a resonant frequency identification method includes broadcasting a broadcast sound toward a predetermined space (SP), measuring sound pressure of the broadcast sound broadcasted toward the predetermined space (SP), and identifying that a resonant frequency in the predetermined space (SP) exists within a specific frequency band, if a change of the measured sound pressure in the specific frequency band is different from a change of sound pressure in a case where there is no resonance in the predetermined space (SP).

An IoT Smart Sound Level Meter that is connected through the internet via ethernet cable, wifi or cellular network. The IoT Smart Sound Level Meter measures and monitors sound pressure levels. Sound pressure levels are measured in decibels. The sound pressure level data that is sent via the internet through ethernet, wifi or cellular network to cloud servers where the data is stored and saved. The cloud servers are connected to Application Programming Interfaces (API) connected to web-based software that converts the sound pressure level data into decibel measurements. These measurements are displayed to the end-user through the web-based software and / or application providing real-time decibel level measurements and historic sound pressure level recordings. The web-based software and / or application provides access to view decibel measurement data remotely from a computer, laptop, tablet, smart watch, or mobile device.

An apparatus detects complex time- variant electromagnetic disturbances resulting from a high- altitude nuclear electromagnetic pulse (EMP) or solar storm. The device relies on a circuit (104, 1301) to monitor the input power lines (301, 401, 501) for sustained conducted electrical disturbances associated with the E3 phase of an EMP or solar storm. A separate circuit (105, 1302) monitors the power lines and the ambient environment (503) for transient electromagnetic pulse disturbances associated with the El and E2 phases of an EMP. When sustained electrical disturbances or transient electromagnetic pulse disturbances are detected, the apparatus provides a visual alarm (603. 609), audible alarm (608), and discrete indication signal (607) that can be used to disconnect (702) or redirect the flow of electrical power.

A method for reproduction of at least one reflection of reverberation in virtual acoustics rendering systems, the method comprising: obtaining for a room, a geometry of the room and at least one spatial room impulse response, wherein the at least one spatial room impulse response is obtained based on a measurement in the room; determining a reverberation time parameter based on the at least one spatial room impulse response; determining a room absorption area based on the determined reverberation time parameter and the geometry of the room; determining at least one parameter of early reflections from the at least one spatial room impulse response based on at least one image source position, wherein the at least one image source position is associated with early reflections of the at least one spatial room impulse response; extracting at least one reflection from the at least one spatial room impulse response; extracting at least one reflection coefficient based on the extracted at least one reflection for at least one surface of the room; assigning absorption coefficients for at least one surface of the room based on the at least one reflection coefficient and the determined room absorption area; and rendering the at least one early reflection based on the determined absorption coefficients.

Audio signal dereverberation may be performed using a processor according to instructions on a machine-readable storage medium. In one example, a position of a person in a room may be determined. Beamforming may be used to capture an audio signal received from the position of the person. Room properties may be determined based in part on a signal sweep of the room. Dedereverberation parameters may be determined based in part on the position of the person and the room properties. The dedereverberation parameters may be applied to the audio signal.