Abstract: Reconstruction method for reconstructing a first signal (x(t)) regularly sampled at a sub-Nyquist rate, comprising the step of retrieving from the regularly spaced sampled values (ys[n], y(nT)) a set of weights (cn, cnr, ck) and shifts (tn, tk) with which said first signal (x(t)) can be reconstructed. The reconstructed signal (x(t)) can be represented as a sequence of known functions (?(t)) weighted by the weights (ck) and shifted by the shifts (tk). The sampling rate is at least equal to the rate of innovation (?) of the first signal (x(t)).

Abstract: The aim of the invention is to provide inter-channel level differences ICLD related to audio signals for hearing aids.

Abstract: Method for decoding a signal sent over a bandwidth-expanding communication system, where both channel estimation and signal detection are carried out on a set of samples generated by sampling the received signal at a sub-Nyquist rate, thus allowing for a significant reduction of the complexity of the sampling device of receivers using said method, as well as a significant reduction of their computational requirements.

Abstract: Reconstruction method for reconstructing a first signal (x(t)) regularly sampled at a sub-Nyquist rate, comprising the step of retrieving from the regularly spaced sampled values (ys[n], y(nT)) a set of weights (cn, cnr, ck) and shifts (tn, tk) with which said first signal (x(t)) can be reconstructed. The reconstructed signal (x(t)) can be represented as a sequence of known functions (?(t)) weighted by the weights (ck) and shifted by the shifts (tk). The sampling rate is at least equal to the rate of innovation (?) of the first signal (x(t)).

Abstract: A reconstruction method for reconstructing a first signal from a set of sampled values generated by sampling a second signal at a sub-Nyquist rate and at uniform intervals, the method includes retrieving from the set of sampled values a set of shifts and weights with which the first signal can be reconstructed.

Abstract: A method for image acquisition and conversion includes low-pass filtering an image by an acquisition lens, producing from the low-pass filtered image, an up-sampled image with a first resolution with an up-sampling factor using a image sensor. The up-sampled image is converted into a multi-level image with a second resolution lower than the first resolution with an image processing circuit. The converting depends on the low-pass filtering of the lens and on the up-sampling factor. The method is adapted to gigapixel sensors and convention image sensors.

Abstract: An iterative method for calibrating a transducer array characterized by calibration parameters, based on time delay measurements between transducer pairs, comprising the steps of: creating a distance data set that includes pairwise distances between estimated positions of transducer; forming a time delay measurement data set that includes time delay measurements between transducer pairs; generating from the time delay measurement data set an estimate of at least one calibration parameter; modifying the time delay measurement data set and the distance data set based on the estimated calibration parameters; denoising the modified distance data set with an iterative algorithm; updating the estimated positions of transducers based on the denoised modified distance data set; and repeating at least a portion of the steps until the difference between the estimated positions of transducer pairs and updated estimated positions of transducers satisfies a stopping criterion.

Abstract: In previously known imaging devices as in still and motion cameras, for example, image sensor signal response typically is linear as a function of intensity of incident light. Desirably, however, akin to the response of the human eye, response is sought to be nonlinear and, more particularly, essentially logarithmic. Preferred nonlinearity is realized in image sensor devices of the invention upon severely limiting the number of pixel states, combined with clustering of pixels into what may be termed as super-pixels.

Abstract: Embodiments include a method for sending a selected number of pilots (20) to a sparse channel having a channel impulse response limited in time comprising sending the selected number of the pilots (20). The pilots (20) are equally spaced in the frequency domain the number is selected based on the finite rate of innovation of the channel impulse response. Once received the pilots (20), such a channel is estimated by: low-pass filtering (100) the received pilots, sampling (200) the filtered pilots with a rate below the Nyquist rate of the pilots, applying a FFT (300) on the sampled pilots, verifying (500) the level of noise of the transformed pilots, if the level of noise is below to a determined threshold, applying an annihilating filter method (600) to the transformed pilots, and dividing the temporal parameters by the distance (D) between two consecutive pilots.

Abstract: A method for adaptively learning a sparse impulse response (100) of a continuous channel to which an input signal (x (t)) is applied and which delivers an output signal (y(t)), comprising the following steps: low-pass filtering the input signal and the output signal and obtain a filtered input signal (xF(t)) and a filtered output signal (yF(t)) sampling the filtered input signal and the filtered output signal with a sampling rate below the Nyquist rate and obtaining a sampled input signal (xS(t)) and a sampled output signal (yS(t)) retrieving from the sampled input signal (xS(t)) and the sampled output signal (yS(t)) an estimate (400) of the sparse impulse response (100) of the continuous channel. This method can be applied in CDMA channels, in acoustic room context, in ultra-wideband ranging and line echo cancellation problems, in transmission systems for optical fibres, in body scan devices, to name a few.

Abstract: Given two video sequences, a composite video sequence can be generated which includes visual elements from each of the given sequences, suitably synchronized and represented in a chosen focal plane. For example, given two video sequences with each showing a different contestant individually racing the same down-hill course, the composite sequence can include elements from each of the given sequences to show the contestants as if racing simultaneously. A composite video sequence can be made also by similarly combining a video sequence with an audio sequence.

Abstract: Reconstruction method for reconstructing a first signal (x(t)) regularly sampled at a sub-Nyquist rate, comprising the step of retrieving from the regularly spaced sampled values (ys[n], y(nT)) a set of weights (cn, cnr, ck) and shifts (tn, tk) with which said first signal (x(t)) can be reconstructed. The reconstructed signal (x(t)) can be represented as a sequence of known functions (?(t)) weighted by the weigths (ck) and shifted by the shifts (tk). The sampling rate is at least equal to the rate of innovation (?) of the first signal (x(t)).

Abstract: Certain aspects of the present disclosure provide methods for distributed sensing and centralized reconstruction of two correlated signals, modeled as the input and output of an unknown sparse filtering operation.

Abstract: Reconstruction method for reconstructing a first signal (x(t)) regularly sampled at a sub-Nyquist rate, comprising the step of retrieving from the regularly spaced sampled values (ys[n], y(nT)) a set of weights (cn, cnr, ck) and shifts (tn, tk) with which said first signal (x(t)) can be reconstructed. The reconstructed signal (x(t)) can be represented as a sequence of known functions (?(t)) weighted by the weights (ck) and shifted by the shifts (tk). The sampling rate is at least equal to the rate of innovation (?) of the first signal (x(t)).

Abstract: It is disclosed an acoustic tomography method to improve the time of flight estimation, said method comprising the steps of: sequentially triggering a set of N transmitters so as to generate a sequence of N acoustic waves through a volume being scanned; receiving each of said acoustic waves after transmission through said volume with a set of M receivers, which are called received signals; delaying by varying delays the N different said received signals that each receiver receives from the N different transmitters, and adding them together to form a new received signal, which is called transmit-beamformed signal for that receiver; delaying by varying delays the M different said transmit-beamformed signals for each receiver and adding them together at each receiver to form a new signal, which we call transmit-receive-beamformed signal.

Abstract: An encoder/decoder for multi-channel audio data, and in particular for audio reproduction through wave field synthesis. The encoder comprises a two-dimensional filter-bank to the multi-channel signal, in which the channel index is treated as an independent variable as well as time, and and the resulting spectral coefficient are quantized according to a two-dimensional psychoacoustic model, including masking effect in the spatial frequency as well as in the temporal frequency. The coded spectral data are organized in a bitstream together with side information containing scale factors and Huffman codebook identifiers.

Abstract: Signals, including signals from outside of the subspace of bandlimited signals associated with the Shannon theorem, are acquired while still providing an acceptable reconstruction. In some aspects a denoising process is used in conjunction with sparse sampling techniques. For example, a denoising process utilizing a Cadzow algorithm may be used to reduce the amount of noise associated with sampled information. In some aspects the denoising process may be iterative such that the denoising process is repeated until the samples are denoised to a sufficient degree. In some aspects, the denoising process converts a set of received samples into another set corresponding to a signal with a Finite Rate of Innovation (FRI), or to an approximation of such a signal. The disclosure relates in some aspects to combination of a denoising process with annihilating filter methods to retrieve information from a noisy, sparse sampled signal.

Abstract: Signals, including signals from outside of the subspace of bandlimited signals associated with the Shannon theorem, are acquired while still providing an acceptable reconstruction. In some aspects a denoising process is used in conjunction with sparse sampling techniques. For example, a denoising process utilizing a Cadzow algorithm may be used to reduce the amount of noise associated with sampled information. In some aspects the denoising process may be iterative such that the denoising process is repeated until the samples are denoised to a sufficient degree. In some aspects, the denoising process converts a set of received samples into another set corresponding to a signal with a Finite Rate of Innovation (FRI), or to an approximation of such a signal. The disclosure relates in some aspects to combination of a denoising process with annihilating filter methods to retrieve information from a noisy, sparse sampled signal.

Abstract: The aim of the invention is to provide inter-channel level differences ICLD related to audio signals for hearing aids.

Abstract: Method for decoding a signal sent over a bandwidth-expanding communication system, where both channel estimation and signal detection are carried out on a set of samples generated by sampling the received signal at a sub-Nyquist rate, thus allowing for a significant reduction of the complexity of the sampling device of receivers using said method, as well as a significant reduction of their computational requirements.