69 results about "Acoustic transfer function" patented technology

A surround sound system for reproducing a spatial sound field in a sound control region within a room having at least one sound reflective surface. The system uses multiple steerable loudspeakers located about the sound control region, each loudspeaker having a plurality of different individual directional response channels being controlled by respective speaker input signals to generate sound waves emanating from the loudspeaker with a desired overall directional response. A control unit connected drives each of the loudspeakers and has pre-configured filters based on measured acoustic transfer functions for the room for filtering the input spatial audio signals to generate the speaker input signals for all the loudspeakers to generate sound waves with co-ordinated overall directional responses that combine together at the sound control region in the form of either direct sound or reflected sound from the reflective surface(s) of the room to reproduce the spatial sound field.

The present invention provides a sound outputting apparatus, including: a housing; an electro-acoustic conversion section provided in the housing and configured to acoustically reproduce and output a sound signal; an acousto-electric conversion section provided at a position of the housing at which sound acoustically reproduced by the electro-acoustic conversion section can be collected; a removing section configured to remove a component of the sound signal from an output signal to be outputted from the acousto-electric conversion section based on an acoustic transfer function between the electro-acoustic conversion section and the acousto-electric conversion section; a decision section configured to decide whether or not a predetermined operation is performed for the housing based on an output signal from the removing section; and a control section configured to control so that a predetermined process is performed based on a result of the decision by the decision section.

The invention provides a canal hearing device and method in which the device is implemented with a mode of operation that provides acoustic transparency as well as a power-saving function, particularly useful to permit the user to wear the device in the ear canal during periods of sleep or inactivity without substantial loss of normal unaided response. The transparent mode has an in-situ acoustic transfer function that compensates for the insertion loss caused by the presence of a hearing device in the ear canal. While the device is in this transparent mode, its acoustic transfer function gives the user a perception of unaided hearing, as though the device were removed, when it is actually being worn continuously in the ear canal. Current drain of the device is significantly reduced as the transparent mode serves to shut off or reduce bias currents of at least one circuit element within the device circuitry. The invention is particularly useful in canal hearing devices adapted for extended wear in the ear canal for periods longer than one month without removal.

An apparatus estimates the direction of a sound source from signals plural microphones capture sound to produce. Data are stored on reverse characteristics of spatial transfer functions defined on sound transmitted from sound source positions to the respective microphones. To the signal produced by each microphone, applied are the reverse characteristics of the spatial transfer functions thus stored in connection with that microphone with respect to the sound source positions to thereby estimate a sound source signal on a sound source position associated with the sound captured. Between the sound source signals estimated on the sound source positions associated with the sounds captured by the microphones, coincidence or higher correlation is found on a sound source position to thereby produce information on at least the direction of the sound source thus found.

Embodiments of the invention disclose computer-implemented methods, systems, and computer program products for estimating signal coherence. First, a sound generated by a sound source is detected by a first microphone to obtain a first microphone signal and by a second microphone to obtain a second microphone signal. The first microphone signal is filtered by a first adaptive finite impulse response filter to obtain a first filtered signal. The second microphone signal is filtered by a second adaptive finite impulse response filter, to obtain a second filtered signal. The coherence of the first filtered signal and the second filtered signal is determined based upon the filtered signals. The first and the second microphone signals are filtered such that the difference between the acoustic transfer function for the transfer of the sound from the sound source to the first microphone and the transfer of the sound from the sound source to the second microphone is compensated in the first and second filtered signals.

A sound source location is estimated by particle filtering where a set of particles represents a probability density function for a state variable comprising the sound source location. The method includes determining the weight for a particle in response to a correlation between estimated acoustic transfer functions from the sound source to at least two sound recording positions. A weight update function may specifically be determined deterministically from the correlation and thus the correlation may be used as a pseudo-likelihood function for the measurement function of the particle filtering. The acoustic transfer functions may be determined from an audio beamforming towards the sound source. The audio weight may be combined with a video weight to generate a multi-modal particle filtering approach.

The invention concerns sound spatialization with multichannel encoding for binaural reproduction on two loudspeakers, the spatial encoding being defined by encoding functions associated with multiple encoding channels and the decoding by applying filters for binaural reproduction. The invention provides for an optimization as follows: a) obtaining a original set of acoustic transfer functions particular to an individual's morphology (HRIR;HRTF), b) selecting spatial encoding functions (g(θ,φ,n)) and / or decoding filters (F(t,n)), and c) through successive iterations, optimizing the filters associated with the selected encoding functions or the encoding functions associated with the selected filters, or jointly the selected filters and encoding functions, by minimizing an error (c(HRIR,HRIR*)) calculated based on a comparison between: the original set of transfer functions (HRIR), and a set of reconstructed transfer functions (HRIR*) from encoding functions and decoding filters, whether optimized and / or selected.

Technologies are generally described for automatically managing noise profile in a predefined area by determining a desired noise profile within the predefined area, monitoring noise levels and / or frequencies within the predefined area, and adjusting operational aspects of one or more noise emitting devices in order to achieve the desired noise profile within the predefined area. A noise management system according to embodiments may be centrally controlled or organized in a distributed manner with control modules on individual noise emitting devices interacting through wired or wireless media. Furthermore, the adjustment of the operations of the noise emitting devices in order to achieve the desired noise profile may be accomplished through computing an acoustic transfer function or measuring actual noise levels / frequencies.

The invention provides a far field identification processing method and device. The far field identification processing method includes the steps: acquiring an ATF (acoustic transfer function) of an application environment according to an ATF acquired by an AEC (acoustic echo cancellation) module; acquiring near field identification training data; acquiring far field identification training data corresponding to the application environment according to the ATF of the application environment and the near field identification training data. According to the method, additional participation of a user can be omitted, and interference in the user is reduced, so that user experience is improved.

The invention relates to a method for automated tuning of a sound system, the sound system comprising delay lines, equalizing filters, and at least two loudspeakers, the method comprising the steps of reproducing a useful sound signal through the loudspeakers, measuring sound pressure values at at least one location, providing a target transfer function for tuning the delay lines and the equalizing filters of the sound system, the target transfer function representing a desired transfer characteristics of the sound system, adjusting the delay of the delay lines, and adjusting amplitude responses of the equalizing filters such, that the actual transfer characteristics of the sound system approximates the target function.

Provided is a sound image localization control apparatus for allowing, when sound is reproduced so as to perform sound image localization for a plurality of users, each of the plurality of users to variably adjust an acoustical effect individually without diminishing a sound image localization effect. The sound image localization control apparatus comprises: processing characteristic setting means (13; 14) for setting a processing characteristic in controlling means such that acoustic transfer functions for at least two predetermined positions each represent a desired characteristic; controlling means (12) for receiving an acoustic signal and the processing characteristic which is set by the processing characteristic setting means and performing signal processing; and sound reproducing means (3) for receiving an output from the controlling means.

The invention relates to an audio signal processing apparatus (100) comprising a determiner (101) being configured to determine a filter matrix C on the basis of an acoustic transfer function matrix H and a target acoustic transfer function matrix VH, wherein the acoustic transfer function matrix H comprises transfer functions of acoustic propagation paths between loudspeakers and a listener and the target acoustic transfer function matrix VH comprises target transfer functions of target acoustic propagation paths, wherein the target acoustic propagation paths are defined by a target arrangement of virtual loudspeaker positions relative to the listener, a filter (103) being configured to filter the input audio signal on the basis of the filter matrix C to obtain filtered input audio signals, and a combiner (105) being configured to combine the filtered input audio signals to obtain output audio signals.

The invention provides a compensating filter fitting system, a sound compensation system and methods. The compensating filter fitting system comprises a target system, a simulation system and a fitting module, wherein the target system comprises a first transfer function module and is suitable for outputting target sound signals after input sound signals are processed with a first transfer function in the first transfer function module; the simulation system is connected with the target system, comprises a second transfer function module and is suitable for outputting simulated sound signals after the input target sound signals are processed with a second transfer function in the second transfer function module; the fitting module is connected with the target system and the simulation system and is suitable for fitting the target sound signals and the simulated sound signals, so that parameters of the compensating filter connected with the input end of the simulation system are obtained. By means of acquisition and addition of the compensating filter, distortion of the simulation system can be eliminated, and the target sound signals can be restored with high quality; one simulation system can be used for simulating other different types of sounds with different tone quality laterality.

A three-dimensional array (10) of microphones (M) is used to determine the three-dimensional sound field above a surface element (ΔS) of a sound emitting surface (S), and the air-particle velocity (un) at the surface element (ΔS) is determined using Near-Field Acoustical Holography (NAH). A volume velocity sound source (11) is used to emit a volume velocity (Qv) in a listening position, and the array (10) of microphones (M) is used to determine the resulting three-dimensional sound field above the surface element (ΔS), and using NAH the resulting sound pressure at the surface element (ΔS) is determined. The acoustic transfer function (H) between the surface element (ΔS) and the listening position is assumed to be reciprocal and is determined as the ratio of the resulting sound pressure at the surface element (ΔS) to the volume velocity (Qv). The sound pressure in the listening position resulting from the surface element (ΔS) is determined as Δp=H.(un. ΔS).

The present invention provides a time delay estimation method and device. The method comprises: when detecting that the current frame of a first audio signal is a speech frame, calculating the signal to noise ratio of the current frame; performing adaptive adjustment of a speech signal data block corresponding to the current frame according to the signal to noise ratio of the current frame, wherein the speech signal data block is configured to construct the linear system of equations for estimation of an acoustic transfer functions ratio; and constructing the linear system of equations according to the data block after regulation, solving the estimation value of the acoustic transfer functions ratio, and determining the time delay estimation value according to the estimation value of the acoustic transfer functions ratio. Therefore, the time delay estimation method and device can improve the time delay estimation accuracy.

Methods and apparatus are provided for controlling noise in a compartment. The audio system includes an error microphone configured to receive sounds and generate an error signal corresponding to the received sounds. A processor in communication with the error microphone is configured receive the error signal from the error microphone and generate a noise-canceling signal based at least in part on the error signal and an acoustic transfer function. The audio system also includes a loudspeaker in communication with the processor to receive the noise-canceling signal and produce a noise-canceling sound wave based on the noise-canceling signal. The processor is also configured to receive at least one audio signal different from the error signal and to modify the acoustic transfer function utilizing the at least one audio signal.

The invention discloses a rapid echo cancellation method in an intelligent sound box. The method comprises the following steps: 1, establishing an acoustic function model comprising a voice signal, anecho signal, background noise and a microphone receiving signal; 2, obtaining an acoustic transfer function model of each microphone; 3, obtaining the upper branch voice reference signal of the fixedbeamformer; 4, calculating a first channel echo signal; 5, calculating the relative echo transfer function of the first second echo signal according to the space structure of the microphone array toobtain echo signals of other channels of the microphone array; and 6, constructing a cost function according to the minimum mean square error value between the voice reference signal obtained by fixedbeamforming and the adaptive beamforming noise reference signal to update and iterate a beamformer coefficient. According to the method, echo cancellation is carried out in combination with an adaptive beam forming algorithm, residual echo estimation does not need to be carried out, and small distortion of a target voice signal can be guaranteed while the echo is well suppressed.

The invention discloses an immersion type broadband 3D sound field replay method, which comprises the following steps: firstly, calculating an acoustic transfer function of each audition point from avirtual sound source of a scene A placed in an appointed spatial position to a scene B, wherein the function value is used as a sound pressure value of a radiation sound field of the virtual sound source; secondly, setting a loudspeaker array of a certain wall surface in the scene B at a regular rectangular equal-spacing layout, and modeling acoustic transfer features from all the loudspeakers tothe audition points by using a green function based on a fluctuation characteristic of sound waves; thirdly, based on the linear convex optimization theory, performing regularization operation by using an alternate direction multiplier method by taking a l1 norm as a sparse rule operator, and calculating loudspeaker weight values by selecting center frequencies of eight frequency bands within a doubled frequency interval so as to select an activated loudspeaker; and finally, calculating a weight value signal of the activated loudspeaker in a replay system by using l2 norm regularization so asto enable a radiation sound field of a sound source to be replayed to be the most closest to the radiation sound fields of activated loudspeaker under the minimum root-mean-square error criterion.

With the noise reduction device (300), in identifying an acoustic transfer function that includes a path from the control speaker (340) to the error microphone (350) or the noise microphone (320) by outputting an identification sound from the control speaker (340) and detecting the identification sound with the error microphone (350) or the noise microphone (320), the identification controller (338) is configured to identify the acoustic transfer function by generating the identification sound from the white noise generator (337) when the seat detector (582) has detected that the seat is in the actual usage state for noise reduction.

An environmental monitoring device that includes an acoustic sensor is described. During operation of the environmental monitoring device, the acoustic sensor provides acoustic data based on measurements of sound in an external environment that includes the environmental monitoring device. Based on the acoustic data and predefined characterization of the external environment, a control mechanism determines if an alarm device, which is separate from the environmental monitoring device, is activated. For example, the predefined characterization may include a location of the alarm device. This location may be specified by: an image of the external environment, a positioning system, a communication network, and / or an acoustic latency in the external environment. Alternatively, the predefined characterization may include an acoustic transfer function of the external environment proximate to the alarm device and the environmental monitoring device. Furthermore, if the alarm device is activated, the environmental monitoring device may provide an alert.