5502 results about "Transfer function" patented technology

An audio privacy system reduces the intelligibility of speech in an audio signal while preserving prosodic information, such as pitch, relative energy and intonation so that a listener has the ability to recognize environmental sounds but not the speech itself. An audio signal is processed to separate non-vocalic information, such as pitch and relative energy of speech, from vocalic regions, after which syllables are identified within the vocalic regions. Representations of the vocalic regions are computed to produce a vocal tract transfer function and an excitation. The vocal tract transfer function for each syllable is then replaced with the vocal tract transfer function from another prerecorded vocalic sound. In one aspect, the identity of the replacement vocalic sound is independent of the identity of the syllable being replaced. A modified audio signal is then synthesized with the original prosodic information and the modified vocal tract transfer function to produce unintelligible speech that preserves the pitch and energy of the speech as well as environmental sounds.

Active noise control system and method for controlling an acoustic noise generated by a noise source at a listening location, in which system and method sound is picked up in the surroundings of the listening location by a sound sensor; an electrical noise signal which corresponds to the acoustic noise of the noise source is generated and filtered adaptively in accordance with control signals. The adaptively filtered noise signal is irradiated into the surroundings of the listening location by a sound reproduction device, where a secondary path transfer function extends between the sound reproduction device and sound sensor. The noise signal is filtered with a transfer function that models the secondary path transfer function. The signals which provided by the sound sensor after first filtering serve as control signals for the adaptive filtering.

A system and method for protecting data within a portable electronic device to prevent unauthorized access to that data. Encryption of data within the portable electronic device and automatic erasure of data upon unauthorized attempted access is provided. A limited number of attempts to access the portable electronic device are allowed, and if exceeded, the data is automatically erased without notification. Data transfer functions of the portable electronic device are also disabled when the device is locked. Erasing of data is also provided if the portable electronic device is not synced with another device during a predetermined time period.

A system for providing an accurate prediction of a signal-to-interference noise ratio is described. The system includes a first circuit for receiving a signal transmitted across a channel via an external transmitter. A second circuit generates a sequence of estimates of signal-to-interference noise ratio based on the received signal. A third circuit determines a relationship between elements of the sequence of estimates. A fourth circuit employs the relationship to provide a signal-to-interference noise ratio prediction for a subsequently received signal. In the illustrative embodiment, the inventive system further includes a circuit for generating a data rate request message based on the signal-to-noise ratio prediction. A special transmitter transmits the data rate request message to the external transmitter. In the specific embodiment, the relationship between elements of the sequence of estimates is based on an average of the elements of the sequence of estimates. The third circuit includes a bank of filters for computing the average. The bank of filters includes finite impulse response filters. Coefficients of the transfer functions associated with each filter in the bank of filters are tailored for different fading environments. The different fading environments include different Rayleigh fading environments, one environment associated with a rapidly moving system, a second environment associated with a slow moving system, and a third system associated with a system moving at a medium velocity. A selection circuit is connected to each of the filter banks and selects an output from one of the filters in the filter bank. The selected output is associated with a filter having a transfer function most suitable to a current fading environment.

Digital focusing method and apparatus in an image processing system, for digitally focusing an out-of-focus image, are provided. A defocused image is divided into sub-images of a predetermined size. An edge direction of each of the divided sub-images is estimated. Step responses with respect to the respective edge directions are calculated. A mean step response is obtained by averaging a predetermined number of the step responses. Point Spread Function (PSF) coefficients are obtained using the mean step response. An image blur transfer function is obtained using the PSF coefficients. An image restoration transfer function is obtained using the image blur transfer function. An original in-focused image is obtained by multiplying the image restoration transfer function by the defocused image in a frequency domain. Thus, an image can be restored in real time, and the size and weight of the image processing system can be reduced.

An insertion-based error concealment method and apparatus are provided whereby, instead of directly inserting white noise, a filter is created to shape the white noise. The filtered white noise is then used to replace lost data. The method of the present invention is implemented by first estimating the power spectrum of the previous frame; then designing a filter with transfer function H(f), where |H(f)|²=the estimated power spectrum; and finally generating the replacement packet using noise which has been spectrally modified by the filter. The resulting filtered noise has the same power spectrum as the previous packet but is not highly correlated with it.

A method and computer-readable medium are provided for identifying clean signal feature vectors from noisy signal feature vectors. The method is based on variational inference techniques. One aspect of the invention includes using an iterative approach to identify the clean signal feature vector. Another aspect of the invention includes using the variance of a set of noise feature vectors and/or channel distortion feature vectors when identifying the clean signal feature vectors. Further aspects of the invention use mixtures of distributions of noise feature vectors and/or channel distortion feature vectors when identifying the clean signal feature vectors. Additional aspects of the invention include using a variance for the noisy signal feature vector conditioned on fixed values of noise, channel transfer function, and clean speech, when identifying the clean signal feature vector.

A system is provided for configuring an audio system for a given space. The system may statistically analyze potential configurations of the audio system to configure the audio system. The potential configurations may include positions of the loudspeakers, numbers of loudspeakers, types of loudspeakers, listening positions, correction factors, filters, or any combination thereof. The statistical analysis may indicate at least one metric of the potential configuration including indicating consistency of predicted transfer functions, flatness of the predicted transfer functions, differences in overall sound pressure level from seat to seat for the predicted transfer functions, efficiency of the predicted transfer functions, or the output of predicted transfer functions. The system also provides a methodology for selecting loudspeaker locations, the number of loudspeakers, the types of loudspeakers, correction factors, listening positions, crossover filters or a combination of these schemes in an audio system that has a single listening position or multiple listening positions.

Intelligibility of a human voice projected by a loudspeaker in an environment of high ambient noise is enhanced by processing a voice signal in accordance with the frequency response characteristics of the human hearing system. Intelligibility of the human voice is derived largely from the pattern of frequency distribution of voice sounds, such as formants, as perceived by the human hearing system. Intelligibility of speech in a voice signal is enhanced by filtering and expanding the voice signal with a transfer function that approximates an inverse of equal loudness contours for tones in a frontal sound field for humans of average hearing acuity.

A component that functions with bulk acoustic waves, particularly a bandpass filter, has an increased number of degrees of design freedom in order to improve the transmission characteristics of the component. The component has BAW resonators coupled acoustically in the vertical and/or lateral direction through common electrodes, coupling layer systems and through the excitation of lateral acoustic modes. Through the acoustic coupling of the resonators, it is possible to create additional pole points in the transfer function so that in this manner, the rejection band characteristics of a bandpass filter can be improved. Through acoustic paths which are added in addition to the electrical connection, the insertion loss can be reduced. Through an acoustic coupling instead of an electrical connection, decoupling between input and output loops of a circuit can be achieved.

A single frequency relay station receives a signal from a master station, and retransmits it. Before retransmitting the signal, the relay station uses a digital filter to subtract from the signal components which arise from multipath interference and coupling interference. The coefficients used by the digital filter are derived from characteristics of the multipath interference and the overall transfer function of the relay station. These are derived by turning off the retransmission, so that the multipath interference can be estimated from the received signal, and commencing the retransmission again, to determine the transfer function. A simplified inverse fast Fourier transform is used to simplify the calculations.

A noise canceling and communication system is described. An in-ear device is adapted to fit in the ear canal of a device user. A passive noise reduction element reduces external noise entering the ear canal. An external microphone senses an external acoustic signal outside the ear canal to produce a representative external microphone signal. An internal microphone senses an internal acoustic signal proximal to the tympanic membrane to produce a representative internal microphone signal. One or more internal sound generators produce a noise cancellation signal and an acoustic communication signal, both directed towards the tympanic membrane. A probe tube shapes an acoustic response between the internal sound generator and the internal microphone to be relatively constant over a wide audio frequency band. An electronics module is located externally of the ear canal and in communication with the in-ear device for processing the microphone signals using a hybrid feed forward and feedback active noise reduction algorithm to produce the noise cancellation signal. The noise reduction algorithm includes a modeling component based on a transfer function associated with the internal sound generator and at least one of the microphones to automatically adjust the noise cancellation signal for fit and geometry of the ear canal of the user. The communication component also includes a modeling component based on a transfer function associated with the internal sound generator and at least one of the microphones to automatically adjust the communication signal for fit and geometry of the ear canal of the user and to assure that the communication signal does not interfere with the noise reduction algorithm and that the noise cancellation signal does not interfere with passing of the communication signal.

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.

The present invention relates to a blind adaptive filter for narrowband interference cancellation, which includes an adaptive filter, a delay unit coupled to the adaptive filter for generating a delayed signal with a predetermined delay length from the output signal of the adaptive filter, and an error calculation unit coupled to the adaptive filter and the delay unit. The error calculation unit compares the output signal from the adaptive filter and the delayed signal from the delay unit to extract error information, and feedback the first error information to the adaptive filter. The first error information is formed of a transfer function including a number of coefficients, and used to adjust the adaptive filter and remove interference in the next input signal. The disclosed technique is also applicable in wideband receivers, as well as resisting multiple strong narrowband interferences having a frequency sweep rate of tens of milliseconds.

A method is provided for an equalization strategy for compensating channel distortions in a dual-polarization optical transport system wherein the received signal includes a complex signal of a first transmitted polarization component and a complex signal of a second transmitted polarization component. In a first step, a blind self-recovery mode used a blind adaptation algorithm in calculating and modifying multiple complex equalizer transfer function coefficients to enable recovery of only the complex signal of the first transmitted polarization component. By recovering only a single polarization component in the first step the degenerate case of recovering only a single transmitted signal at both polarization component outputs of an equalizer is prevented. In a second step, equalization is performed in a training mode for calculating and modifying the multiple complex equalizer transfer function coefficients to enable recovery of the complex signals of the first and second transmitted polarization components. In a third step, equalization is performed in a data directed mode for continuing to calculate and modify the multiple complex equalizer transfer function coefficients to ensure continued recovery of the complex signals of the first and second transmitted polarization components. The method is suited for a digital signal processing implementation in a coherent receiver when a modulation scheme used on a transmitted signal is quadriphase-shift keying (QPSK). In other embodiments, the method can be used with modulation schemes such as binary PSK, M-ary PSK where M>4, or Quadrature Amplitude Modulation (QAM).

This invention is a method for binaural localization using a cascade of resonators and anti-resonators to implement an HRTF (head-related transfer function). The spectrum of the cascade reproduces the magnitude spectrum of a desired HRTF. The proposed method provides a considerably more computationally efficient implementation of HRTF filters with no detectable deterioration of output quality while saving memory when storing a large quantity of HRTFs due to the parameterization of its resonators and anti-resonators. Finally, the method offers additional flexibility since the resonators and anti-resonators can be manipulated individually during the design process, making it possible to interpolate smoothly between HRTFs, reduce spectral coloring or achieve higher accuracy at perceptually relevant frequency regions. These HRTF are useful in stereo enhancement and multi-channel virtual surround simulation.

A disk drive is disclosed comprising a voice coil motor (VCM) driven in a PWM mode using model reference current feedback. The PWM circuitry and VCM form a plant transfer function which varies with changes to the plant characteristics, such as the resistance of the voice coil fluctuating with temperature drift. A plant model having a model transfer function generates an estimated state of the VCM in response to a detected current flowing through the voice coil. A correction block, responsive to the detected current, adjusts PWM timing signals so that the plant transfer function substantially matches the model transfer function.

A method and an apparatus for transforming a high dynamic range image into a low dynamic range image. The method includes converting first luminance values associated with pixels into a plurality of second luminance values, and utilizing a film transfer function for mapping the second luminance values associated with the pixels into a plurality of third luminance values to generate the low dynamic range image. A second luminance range of the second luminance values is smaller than a first luminance range of the first luminance values, and the film transfer function adds no visual artifact to the low dynamic range image.

An analog-to-digital converter system [50D] processing an input signal, g, which can be either a discrete-time or a continuous-time signal. A first quantizer [154] generates a first digital signal, d0(k), representing the sum of the input signal, g, and a dithering signal, y0. A digital-to-analog converter [156] generates an analog feedback signal, alpha, representing accurately the first digital signal, d0(k). The DAC [156] may be linearized by the use of mismatch-shaping techniques. A filter [158] generates the dithering signal, y0, by selectively amplifying in the signal band the residue signal, r0, defined as the difference of the input signal, g, and the analog feedback signal, alpha. Optional signal paths [166][168] are used to minimize the closed-loop signal transfer function from g to y0, which ideally will be zero. An analog compensation signal, m0, which is described by a well-controlled relationship to the residue signal, r0, is extracted from the filter [158]. Ideally, the closed-loop signal transfer function from g to m0 will be zero, or at least small in the signal band. A second quantizer [160] converts the analog compensation signal, m0, into a second digital signal, dm0(k). The two digital signals, d0(k) and dm0(k), are filtered individually and then added to form the overall output signal, dg(k). The second digital filter [164] has a low signal-band gain, which implies that the sensitivity to signal-band errors caused by the second quantizer [160] will be low. The output signal, dg(k), is a highly-accurate high-resolution representation of the input signal, g. Circuit imperfections, such as mismatch, gain errors, and nonlinearities, will cause only noise-like errors having a very low spectral power density in the signal band.The invention facilitates the implementation of uncalibrated highly-linear high-resolution wide-bandwidth A/D converters [50D], e.g., for use in digital communication systems, such as xDSL modems and other demanding consumer-market products for which low cost is of the essence.

An audio system comprises a receiver (301) for receiving an audio signal, such as an audio object or a signal of a channel of a spatial multi-channel signal. A binaural circuit (303) generates a binaural output signal by processing the audio signal. The processing is representative of a binaural transfer function providing a virtual sound source position for the audio signal. A measurement circuit (307) generating measurement data indicative of a characteristic of the acoustic environment and a determining circuit (311) determines an acoustic environment parameter in response to the measurement data. The acoustic environment parameter may typically be a reverberation parameter, such as a reverberation time. An adaptation circuit (313) adapts the binaural transfer function in response to the acoustic environment parameter. For example, the adaptation may modify a reverberation parameter to more closely resemble the reverberation characteristics of the acoustic environment.

An apparatus for generating a binaural audio signal includes a de-multiplexer and decoder which receives audio data comprising an audio M-channel audio signal which is a downmix of an N-channel audio signal and spatial parameter data for upmixing the M-channel audio signal to the N-channel audio signal. A conversion processor converts spatial parameters of the spatial parameter data into first binaural parameters in response to at least one binaural perceptual transfer function. A matrix processor converts the M-channel audio signal into a first stereo signal in response to the first binaural parameters. A stereo filter generates the binaural audio signal by filtering the first stereo signal. The filter coefficients for the stereo filter are determined in response to the at least one binaural perceptual transfer function by a coefficient processor. The combination of parameter conversion/processing and filtering allows a high quality binaural signal to be generated with low complexity.

An active noise cancellation system reduces, at a listening position, the power of a noise signal being radiated from a noise source to the listening position. The system includes an adaptive filter that receives a reference signal representing the noise signal, and provides a compensation signal. A bass management unit receives the compensation signal and applies a phase shift to the compensation signal to provide a phase shifted compensation signal. A first acoustic radiator receives the phase shifted compensation signal and radiates audio indicative thereof to the listening position. A second acoustic radiator receives the compensation signal and radiates audio indicative thereof to the listening position. The transfer function characteristics from the input of the bass management system to the listening position approximately matches a desired transfer function.