1955 results about "Impulse response" patented technology

A listener can experience the sound of virtual loudspeakers over headphones with a level of realism that is difficult to distinguish from the real loudspeaker experience. Sets of personalized room impulse responses (PRIRs) are acquired for the loudspeaker sound sources over a limited number of listener head positions. The PRIRs are then used to transform an audio signal for the loudspeakers into a virtualized output for the headphones. Basing the transformation on the listener's head position, the system can adjust the transformation so that the virtual loudspeakers appear not to move as the listener moves the head.

A spatial processing stereo system (“SPSS”) that receives audio signals and a limited number of user input parameters associated with the spatial attributes of a room, such as “room size”, “stage distance”, and “stage width”. The input parameters are used to define a listening room and generate coefficients, room impulse responses, and scaling factors that are used generate additional surround signals.

The invention relates to an automated estimation of the position (co-ordinates) of a set of loudspeakers in a ioom Based on measured impulse responses the distances between each pair of loudspeakers are estimated, thereby forming a distance matrix, and the resultant distance matrix is used by a multidimensional scaling (MDS) algorithm to estimate the co-ordinates of each individual loudspeaker An improved co-ordinate estimation can, if desired, be derived by utilizing the stress values provided by the MDS algorithm.

A scheme for stereo and multi-channel synthesis of inter-channel correlation (ICC) (normalized cross-correlation) cues for parametric stereo and multi-channel coding. The scheme synthesizes ICC cues such that they approximate those of the original. For that purpose, diffuse audio channels are generated and mixed with the transmitted combined (e.g., sum) signal(s). The diffuse audio channels are preferably generated using relatively long filters with exponentially decaying Gaussian impulse responses. Such impulse responses generate diffuse sound similar to late reverberation. An alternative implementation for reduced computational complexity is proposed, where inter-channel level difference (ICLD), inter-channel time difference (ICTD), and ICC synthesis are all carried out in the domain of a single short-time Fourier transform (STFT), including the filtering for diffuse sound generation.

A vibration data collection system performs an integration or differentiation process on incoming digitized vibration data in real time. The system uses a digital Infinite Impulse Response (IIR) filter running at the input data rate to provide the integration or differentiation function. With this approach, the system reduces hardware complexity and data storage requirements. Also, the system provides the ability to directly integrate or differentiate stored time waveforms without resorting to FFT processing methods.

A method and apparatus for mapping a wire network is disclosed. The method includes obtaining a reflectometry test signal of the wire network. An estimated network impulse response is estimated from the reflectometry response. A wire network model is then initialized, and iteratively improved by simulating an impulse response of the wire network model and adjusting the wire network model to reduce differences between the simulated impulse response and estimated network impulse response.

A neural network is provided for recognition and enhancement of multi-channel sound signals received by multiple microphones, which need not be aligned in a linear array in a given environment. Directions and distances of sound sources may also be detected by the neural network without the need for a beamformer connected to the microphones. The neural network may be trained by knowledge gained from free-field array impulse responses obtained in an anechoic chamber, array impulse responses that model simulated environments of different reverberation times, and array impulse responses obtained in actual environments.

A calibration method for a two-port VNA includes presenting a high reflection calibration standard and measuring reflection data for each of the two ports, calculating a location of the high reflection calibration standard at each of the two ports, presenting a load calibration standard and measuring the reflection characteristic for each of the two ports to provide load data, converting the load data to the time domain to provide time domain impulse response load data, and gating the time domain impulse response load data based on the locations of the high reflection calibration standard at each of two ports. The method further includes reconstructing frequency domain load data from the gated time domain data, connecting the two ports together and determining forward and reverse transmission characteristics, and calculating systematic error coefficients for the VNA based on the reconstructed frequency domain data and the forward and reverse transmission characteristics.

A methodology for process modeling and control and the software system implementation of this methodology, which includes a rigorous, nonlinear process simulation model, the generation of appropriate linear models derived from the rigorous model, and an adaptive, linear model predictive controller (MPC) that utilizes the derived linear models. A state space, multivariable, model predictive controller (MPC) is the preferred choice for the MPC since the nonlinear simulation model is analytically translated into a set of linear state equations and thus simplifies the translation of the linearized simulation equations to the modeling format required by the controller. Various other MPC modeling forms such as transfer functions, impulse response coefficients, and step response coefficients may also be used. The methodology is very general in that any model predictive controller using one of the above modeling forms can be used as the controller. The methodology also includes various modules that improve reliability and performance. For example, there is a data pretreatment module used to pre-process the plant measurements for gross error detection. A data reconciliation and parameter estimation module is then used to correct for instrumentation errors and to adjust model parameters based on current operating conditions. The full-order state space model can be reduced by the order reduction module to obtain fewer states for the controller model. Automated MPC tuning is also provided to improve control performance.

Multiple Steiner codes are transmitted as bursts from multiple base stations (182, 184, 186) having one or more transmit elements (174, 176, 178, 180), with successive bursts providing an extended training sequence for use in channel estimation at an addressed unit (172), such as a mobile handset. Accurate channel estimation is possible through the use of Wiener frequency domain MMSE deconvolution (518) combined with frequency domain spatial decoupling matrices, with quasi-orthogonal pseudo-noise sequences (502, 504, 520, 522) allocated to base stations and their antenna elements. The use of Steiner codes to supplement Wiener frequency domain MMSE deconvolution and frequency domain spatial decoupling results in the possibility of allocating only a single training sequence to each base station provided that the training sequence is of sufficient length to encompass all multiple time-translated channel impulse responses (H). Estimates may be refined iteratively by minimising the MS error of demodulated pilot symbols. Estimates may also be refined by removing taps from the impulse response which are insignificant based on a relatively long-term power-delay profile for the channel.

A method includes receiving at least two space-time coded signals from an antenna system associated with a first station, determining complex channel state information based on the received space-time coded signals, and sending the complex channel state information to the first station. In an alternative embodiment, a method includes transmitting at least two space-time coded signals in respective beams of a multi-beam antenna array, measuring a channel impulse response for each space-time coded signal at a second station, and sending an indicia of a selected set of least attenuated signals from the second station to the first station. The multi-beam antenna array is associated with a first station. The beams transmit a signature code embedded in each respective space-time coded signal, and the signature codes are orthogonal so that the second station can separate and measure the channel impulse response corresponding to each space-time coded signal.

Embodiments disclosed herein enable detection and improvement of the quality of the audio signal using a mobile device by determining the loss in the audio signal and enhancing audio by streaming the remainder portion of audio. Embodiments disclosed herein enable an improvement in the sound quality rendered by rendering devices by emitting an test audio signal from the source device, measuring the test audio signal using microphones, detecting variation in the frequency response, loudness and timing characteristics using impulse responses and correcting for them. Embodiments disclosed herein also compensate for the noise in the acoustic space by determining the reverberation and ambient noise levels and their frequency characteristics and changing the digital filters and volumes of the source signal to compensate for the varying noise levels.

A channel impulse response is determined from a cross-correlation of a received training sequence and a stored version of the transmitted training sequence. The channel impulse response is iteratively cleansed of noise caused by the finiteness of the cross-correlation. Initial values of the tap weights for the taps of an equalizer such as a decision feedback equalizer may be determined based on the cleansed channel impulse response.

A signal processing system includes an analog-to-digital delta sigma modulator with a duty cycle modulator and a finite impulse response (FIR) filter in a main loop feedback path of the delta sigma modulator. The duty cycle modulator and FIR filter can provide high performance filtering in the main loop feedback path. To prevent instability in the main loop caused by the duty cycle modulator and FIR filter, the delta sigma modulator also includes a stabilizer loop. Transfer functions of the main loop and the stabilizer loop combine to achieve a target transfer function for the analog-to-digital delta sigma modulator that provides for stable operation of the analog-to-digital delta sigma modulator.

Estimates of spectral magnitude and phase are obtained by an estimation process using spectral information from analysis filter banks such as the Modified Discrete Cosine Transform. The estimation process may be implemented by convolution-like operations with impulse responses. Portions of the impulse responses may be selected for use in the convolution-like operations to trade off between computational complexity and estimation accuracy. Mathematical derivations of analytical expressions for filter structures and impulse responses are disclosed.

A novel and useful apparatus for and method of determining the channel order and channel estimate in a communications system such as a wireless communication system including cellular and cordless. Such channels are typically characterized by rapidly changing impulse response and their taps can be modeled as zero-mean, complex, Gaussian random processes. A sufficiently long, initial channel estimate of length is performed so as to ensure that the actual channel taps will be contained in the estimated taps thus making certain that the equalizer will effectively eliminate intersymbol interference. Channel estimation is performed during each burst using the training sequence transmitted in the middle of the burst. The tap energies are averaged so as to track slow variations in the pattern of resultant channel taps. A noise floor is calculated using the lowest averaged taps and a threshold is computed based thereon and applied to the average taps. The channel order and the tap positions are then selected in accordance those average taps that are above the threshold.

The invention relates to a method of mapping subsurface resistivity contrasts by making multichannel transient electromagnetic (MTEM) measurements on or near the earth's surface using at least one source, receiving means for measuring the system response and at least one receiver for measuring the resultant earth response. All signals from the or each source-receiver pair are processed to recover the corresponding electromagnetic impulse response of the earth and such impulse responses, or any transformation of such impulse responses, are displayed to create a subsurface representation of resistivity contrasts. The invention enables subsurface fluid deposits to be located and identified and the movement of such fluids to be monitored.

A receiver circuit has an equalizer that equalizes a received signal propagating through a transmission medium; a data detection circuit that detects an analog output signal of the equalizer at a data sample timing and outputs a digital signal; an intersymbol interference detection circuit that detects an intersymbol interference level from the analog output signal of the equalizer at the data sample timing and from the digital signal of the data detection circuit; and an equalization characteristic control unit that controls the characteristic of the equalizer to minimize the detected intersymbol interference level. The receiver circuit further has a data sample timing control unit in which the data sample timing is controlled to a sample timing at which the difference between the amplitude of the analog output waveform of the equalizer with respect to an impulse and the amplitude of an ideal impulse response waveform is minimal.

The invention discloses a method for testing the assembly performance of a rotor of an aircraft engine, which comprises the following steps of: firstly exciting and vibrating a rotor of an aircraft engine with a vibration exciter; obtaining a multiple carrier-coupled impulse response signal of the rotor of the aircraft engine with a vibrating sensor and signal-acquiring system software; analyzing the obtained multiple carrier-coupled impulse response signal of the rotor of the aircraft engine by means of dual-tree complex wavelet transform to obtain eight signal carrier-coupled impulse response signals of the rotor of the aircraft engine; and distilling the average assembly performance index of the obtained eight signal carrier-coupled impulse response signals of the rotor of the aircraft engine, wherein the assembly performance of the rotor of the aircraft engine is judged to be qualified if the obtained average assembly performance index is larger than or equal to 10, and the assembly performance of the rotor of the aircraft engine is judged not to be qualified if the obtained average assembly performance index is less than 10, so that the rotor needs to be repaired.

A photoacoustic probe for port wine stain (PWS), burn and melanin depth measurements is comprised of optical fibers for laser light delivery and a piezoelectric element for acoustic detection. The probe induced and measured photoacoustic waves in acryl amide tissue phantoms and PWS skin in vivo. Acoustic waves were denoised using spline wavelet transforms, then deconvolved with the impulse response of the probe to yield initial subsurface pressure distributions in phantoms and skin. The waves were then analyzed for epidermal melanin concentration, using a photoacoustic melanin index (PAMI) related to the amount of laser energy absorbed by melanin. Propagation time of the photoacoustic wave was used to determine the depth of blood perfusion underlying necrotic, burned tissue. Thus, the photoacoustic probe can be used for determining PWS, burn and melanin depth for most patients receiving laser therapy.

An earpiece for providing an audio signal to two ears of a user, includes an audio interface for providing the audio signal, a bone conduction speaker coupled to the audio interface for outputting an output signal, an adaptive filter with an adjustable impulse response to provide a filtered signal that once played to the second ear of a user, by an in-ear speaker, reduces an audio effect of the leakage signal on the second ear of the user, an in-ear speaker coupled to the audio interface for providing an in-ear audio signal to the second ear of the user, in response to the filtered signal and the second input signal, an in-ear microphone for sensing a sensed audio signal, and a processor for updating an impulse response of the adaptive filter in order to reduce a contribution of the leakage signal to the sensed audio signal.

A telemetric contextually based spatial audio system is integrated into a mobile terminal including a headset and antenna. The headset includes an electronic compass which determines head orientation data for the user. The terminal's geographical position can either be determined locally (e.g. onboard GPS) or by network triangulation. The orientation and position data are sent to a server which overlays the position and head orientation data onto a locally stored “floor plan”, describing an object(s) of interest. The floor plan relates the user's viewpoint to a database of contextual audio clips. A regionally significant portion of the database and respective triggers are sent to the terminal. The terminal using the database information recalculates which audio clip(s) to play. The terminal then convolves the respective audio clip(s) via an impulse response function so as to be perceived by the user as emanating from the exact spatial position of the object specified by the system.

Methods and apparatuses for estimation of a state-of-health in a battery are disclosed. An example method comprises: determining if a terminal voltage of the chemical battery differs from a calculated terminal voltage; in response to determining that the terminal voltage of the chemical battery differs from the calculated terminal voltage, calculating a range of voltages by taking the convolution of a terminal current of the chemical battery with a range of impulse responses from a look up table of impulse responses corresponding to different states-of-health; comparing the terminal voltage of the chemical battery with the range of calculated voltages to determine a second impulse response that corresponds to the case where the terminal voltage matches the calculated voltage; and using the look up table of impulse responses corresponding to different states-of-health to determine the state-of-health of the chemical battery from the second impulse response. Other embodiments are described and claimed.

In described embodiments, a transceiver includes a baud-rate clock and data recovery (CDR) module with an eye sampler, and an adaptation module for adaptively setting parameters of various circuit elements, such as timing, equalizer and gain elements. Data sampling clock phase of the CDR module is set for sampling at, for example, near the center of a data eye detected by the eye sampler, and the phase of data error sampling latch(es) is skewed by the CDR module with respect to the phase of the data sampling latch. Since the error signal driving the timing adaptation contains the information of the pulse response that the CDR module encounters, the phase of timing error sampling latch(es) of the CDR module is skewed based on maintaining a relative equivalence of input pulse response residual pre-cursor and residual post-cursor with respect to the timing error sampling clock phase.

A telemetric contextually based spatial audio system is integrated into a mobile terminal including a headset and antenna. The headset includes an electronic compass which determines head orientation data for the user. The terminal's geographical position can either be determined locally (e.g. onboard GPS) or by network triangulation. The orientation and position data are sent to a server which overlays the position and head orientation data onto a locally stored “floor plan”, describing an object(s) of interest. The floor plan relates the user's viewpoint to a database of contextual audio clips. A regionally significant portion of the database and respective triggers are sent to the terminal. The terminal using the database information recalculates which audio clip(s) to play. The terminal then convolves the respective audio clip(s) via an impulse response function so as to be perceived by the user as emanating from the exact spatial position of the object specified by the system.

A method for carrying out channel equalization in a radio receiver wherein an impulse response is estimated, noise power is determined by estimating a co-variance matrix of the noise contained in a received signal before prefiltering, and tap coefficients of prefilters and an equalizer are calculated. The method comprises determining the noise power after prefiltering by estimating a noise covariance matrix, after which input signals of the channel equalizer are weighted by weighting coefficients obtained from the noise covariance estimation.

A scheme to design an audio precompensation controller for a multichannel audio system, with a prescribed number N of loudspeakers in prescribed positions so that listeners positioned in any of P>1 spatially extended listening regions should be given the illusion of being in another acoustic environment that has L sound sources located at prescribed positions in a prescribed room acoustics. The method provides a unified joint solution to the problems of equalizer design, crossover design, delay and level calibration, sum-response optimization and up-mixing. A multi-input multi-output audio precompensation controller is designed for an associated sound generating system including a limited number of loudspeaker inputs for emulating a number of virtual sound sources. Method includes: estimating, for each loudspeaker input signals, an impulse response at each of a set of measurement positions that cover the P listening regions; specifying a target impulse response (target stages) for each virtual sound source at each measurement position; and determining adjustable filter parameters of the audio precompensation controller so that a criterion function is optimized.

A scheme for stereo and multi-channel synthesis of inter-channel correlation (ICC) (normalized cross-correlation) cues for parametric stereo and multi-channel coding. The scheme synthesizes ICC cues such that they approximate those of the original. For that purpose, diffuse audio channels are generated and mixed with the transmitted combined (e.g., sum) signal(s). The diffuse audio channels are preferably generated using relatively long filters with exponentially decaying Gaussian impulse responses. Such impulse responses generate diffuse sound similar to late reverberation. An alternative implementation for reduced computational complexity is proposed, where inter-channel level difference (ICLD), inter-channel time difference (ICTD), and ICC synthesis are all carried out in the domain of a single short-time Fourier transform (STFT), including the filtering for diffuse sound generation.

The apparatus for determining an impulse response in an environment in which a speaker and a microphone are placed works using an audio signal. Means for spectrally coloring a test signal, which preferably is a pseudonoise signal, works using a psychoacoustic masking threshold of the audio signal to obtain a colored test signal, which is embedded in the audio signal to obtain a measuring signal, which can be fed to the speaker. Means for determining the impulse response preferably performs a cross-correlation of a reaction signal received via the microphone from the environment and the test signal or the colored test signal. With this, an impulse response of an environment may also be determined during the presentation of an audio piece to provide an optimal description of environment for a wave-field synthesis.