27results about "Reverberation time" patented technology

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.

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 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.

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.

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.

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. 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.

Examples of the disclosure relate to enabling external control of noise control processing. In examples of the disclosure a location of a listener is determined. The listener is listening to audio content and the audio content is processed using noise control processing. An acoustic response at the location of the listener is determined and a control signal is provided. The control signal indicates an adjustment for one or more parameters of the noise control processing, and the adjustment is based, at least in part, on the acoustic response at the location of the listener.

Disclosed implementations for determining a reverb characteristic of an environment. An audio signal capturing an amount of sound emitted from a source within an environment over a period of time is received from an audio sensor. A reverb parameter measuring a decrease in the sound over the period of time and a measure of the amount of sound that includes background noise in the environment are determined. The reverb characteristic of the environment is determined based on the reverb parameter and the measure. Spatial audio then rendered based on the reverb characteristic.

A method of supporting arrangement design of acoustic devices receives an acoustic space and a position of a sound source in the acoustic space and a position of a noise source in the acoustic space, calculates, in the received acoustic space, arrangement distribution of the acoustic devices corresponding to the position of the sound source and the position of the noise source that have been received, and displays the calculated arrangement distribution on a display.

The application relates to an environmental noise detection and linkage method of a holder camera, two or more holder cameras integrated with microphone arrays, a road arranged at intervals, and comprising the following steps: S1: the microphone arrays collect information, a controller judges a suspicious sound source, and judges a sound direction according to a direction judgment result of the suspicious sound source; S2: the controller sends a control signal to drive the holder camera to turn to the direction of the suspicious sound source according to the sound direction judged by the microphone arrays; S3: the holder camera locks a target object corresponding to the suspicious sound source according to audio information and recognized image information, records audio and video information, and uploads the audio and video information to a service terminal; and S4: a holder camera at a current position sends a control signal to another holder camera at another position closest to the target object in a moving direction of the target object, so that the holder camera at the other position monitors the target object instead of the holder camera at the current position. The multiple holder cameras are communicated and networked, linkage monitoring is realized, and more accurate judgment can be made.

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).

A method of managing an arrangement of acoustic devices in an acoustic space receives information about the acoustic space, a position of at least one sound source in the acoustic space, and a position of at least one noise source in the acoustic space, calculates, for the acoustic space, the arrangement of the acoustic devices corresponding to the received position of the at least one sound source and the received position of the at least one noise source, wherein the acoustic devices include microphones, and outputs the calculated arrangement of the acoustic devices.

This acoustic processing device (100) is equipped with: an acquisition unit (131) for acquiring the recommended environment which is prescribed for each unit of content and includes the ideal arrangement of speakers in the space where the content is to be played back; a measurement unit (132) for measuring the position of the audience present in the space, the number and arrangement of the speakers, and the shape of the space; and a correction unit (133) which, on the basis of the information measured by the measurement unit, corrects the voice, which is observed from the audience position, of the content produced by the speaker present in the space so as to be a voice produced from a virtual speaker which is ideally positioned in the recommended environment.

A method for reproducing at least one reflection of a reverberation in a virtual acoustic rendering system, the method comprising: for a room, obtaining a geometry of the room and at least one spatial room impulse response, where the at least one spatial room impulse response is obtained based on measurements in the room; determining a reverberation time parameter based on the at least one space room impulse response; determining a room absorption area based on the determined reverberation time parameter and the geometric structure of the room; determining at least one parameter of an early reflection from the at least one spatial room impulse response based on at least one image source location, where the at least one image source location is associated with the early reflection of the at least one spatial room impulse response; extracting at least one reflection from the at least one spatial room impulse response; for at least one surface of the room, extracting at least one reflection coefficient based on the extracted at least one reflection; assigning an absorption coefficient to at least one surface of the room based on the at least one reflection coefficient and the determined room absorption area; and rendering at least one early reflection based on the determined absorption coefficient.

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

A method for reproducing at least one reflection of reverberation in a virtual acoustic rendering system, comprising the steps of: obtaining the shape of a room and at least one spatial room impulse response for a room, wherein the at least one spatial room impulse response is obtained based on measurements in the room; determining reverberation time parameters based on the at least one spatial room impulse response; determining a room absorption area based on the determined reverberation time parameters and the shape of the room; determining at least one parameter of early reflection from the at least one spatial room impulse response based on at least one image source location, wherein the at least one image source location is associated with the early reflection 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 for at least one surface in the room based on the extracted at least one reflection; assigning an absorption coefficient for at least one surface in the room based on the at least one reflection coefficient and the determined room absorption area; and rendering at least one early reflection based on the determined absorption coefficient.

The application provides a reverberation time measurement method, system, device and medium. The method comprises the following steps: calculating the energy value of each frequency band in a sound wave signal to obtain a frequency band energy value; creating a useful sound wave energy model, obtaining an ambient noise energy value of a current environment, inputting the frequency band energy value and the ambient noise energy value into the useful sound wave energy model to obtain a useful sound wave energy value; taking the maximum value of the useful sound wave energy value as a starting point, selecting the useful sound wave energy value in a preset range as a fitting range, fitting the useful sound wave energy value in the fitting range to obtain a fitting straight line; and performing reverberation calculation according to the slope of the fitting straight line to obtain a reverberation time. The application can more accurately estimate the noise power in the test stage, and can also shorten the working time of measurement.

An audio method, system, and computer-readable medium are provided for estimating an acoustic transfer function in an environment between a transducer and a microphone. An audio test signal is generated, and the amplitude or energy of the audio test signal is selected to be at least partially masked by acoustic energy in the environment across a plurality of frequency bins. The audio test signal is provided to a transducer for conversion to an acoustic signal in the environment. A microphone receives the acoustic signal and provides a microphone signal. The microphone signal is processed via a summation method over a predetermined duration of 3 seconds or more, and the acoustic transfer function is estimated based on the summation method.

[Problem] To generate acoustic profile information using a simple operation and without increasing the cost of a multichannel audio device. [Solution] A wireless recording playback terminal 5 is positioned at a listening point of a multichannel audio device 2 as a measuring microphone. The wireless recording playback terminal 5 records, as channel playback data in accordance with a test signal playback command received from the multichannel audio device 2, a test signal outputted from a speaker 3 that corresponds to the channel designated by the command. The wireless recording playback terminal 5 also plays back the sound source of the test signal and records the played-back sound source as sound source playback data without being outputted to the outside from the speaker. An acoustic profile information generation server 4 compares these playback data to measure the delay time and attenuation rate of the channel playback data with respect to the sound source playback data, generates the acoustic profile information of the designated channel on the basis of the result of this measurement, and sets the generated acoustic profile information to the multichannel audio device 2.

The invention relates to a method for determining the direction of an emergency vehicle approaching a vehicle (1). Acoustically specific signals emitted by the emergency vehicle are detected by means of a microphone assembly (2) having a plurality of microphones (2.1 to 2. N) arranged at a distance from one another on the vehicle (1). The microphone assembly (2) is calibrated in a point-specific manner for a respective current position of the vehicle (1) by emitting an acoustic test signal (TS) to the surroundings of the vehicle (1) at the respective current position, detecting a reflection (R) of the test signal (TS) occurring in the surroundings by means of the microphone assembly (2), a sound map (K) is generated on the basis of the reflections (R) and the amplitudes thereof and the directions thereof with respect to the vehicle (1), which sound map indicates the extent to which the emitted test signals (TS) are reflected back to the vehicle (1) from different directions, when determining the emergency vehicle direction on the basis of the sound map (K) generated for the current position, acoustic signals received by means of microphones (2.1 to 2. N) of the microphone assembly (2) are weighted, said microphones being oriented in different directions.

The invention relates to a method for ascertaining an emergency vehicle direction from which an emergency vehicle is approaching a vehicle (1). A special acoustic signal emitted by the emergency vehicle is detected by means of a microphone assembly (2) which has a plurality of microphones (2.1 to 2.n) arranged on the vehicle (1) in a mutually spaced manner. The microphone assembly (2) is calibrated for the respective current position of the vehicle (1) in a location-specific manner in that an acoustic test signal (TS) is emitted into the surroundings of the vehicle (1) at the respective current position, reflections (R) of the test signal (TS) produced in the surroundings are detected by the microphone assembly (2), and a sound map (K) is generated on the basis of the reflections (R), the amplitude thereof, and the direction thereof relative to the vehicle (1), said sound map indicating to what extent the emitted test signal (TS) is reflected back to the vehicle (1) from different directions. Received acoustic signals are weighted when ascertaining the emergency vehicle direction on the basis of the sound map (K) generated for the current position by means of the microphones (2.1 to 2.n) of the microphone assembly (2), said microphones being oriented in different directions.