Ciena Corporation

Sivantos Pte. Ltd.

University of Ottawa

Her Majesty the Queen in Right of Canada As Represented by the Minister of Industry Through the Communications Research Centre Canada

Martin Bouchard

Shahab Oveis Gharan

Kim Bryon Roberts

Eghart Fischer

Henning Puder

Ramin Babaee

Homayoun Kamkar-Parsi

Hala As'ad

Homayoun Kamkar Parsi

Louis Thibault

Frederic Mustiere

Hossein Najaf-Zadeh

Ramin Pishehvar

Hassan Lahdili

Sa'di Altamimi

Hala AS´AD

Aspects of the subject disclosure may include, for example, identifying a modulation scheme for transmitting data from a transmitter to a receiver, identifying a plurality of symbols associated with the modulation scheme, determining that a transfer function of a filter that is applied at the transmitter imposes an interdependence amongst at least two symbols of the plurality of symbols, determining, based on the transfer function, a joint probability distribution of the plurality of symbols that reduces a signal power of at least one signal that is used to transmit the data subject to a constraint that an information rate associated with the at least one signal is greater than a threshold, and causing the data to be transmitted via the at least one signal from the transmitter to the receiver in accordance with the joint probability distribution. Other aspects are disclosed.

Apparatuses and methods for facilitating joint precoding and probabilistic constellation shaping

An apparatus comprises circuitry configured to generate a predicted error signal by applying to a digital signal a distortion model characterized by parameters configured to model circuit component mismatches in a digital-to-analog converter (DAC), circuitry configured to generate a pre-compensated digital signal using the digital signal and the predicted error signal, and circuitry configured to provide the pre-compensated digital signal to the DAC for conversion into an analog signal.

Digital-to-analog converter (DAC) distortion pre-compensation

A digital-to-analog converter (DAC) comprises circuitry configured to generate, based on a mapping, L signals representing an N-bit digital input, wherein N and L are positive integers, and wherein N&#x3c;L&#x3c;2&#x2212;1, and circuitry configured to control current flow from L weighted current sources using the L respective signals, thereby generating an analog output that uniquely represents the N-bit digital input, wherein the weighted current sources have weights configured to minimize at least one error metric associated with the analog output.

Digital-to-analog converter (DAC) architecture optimization

A receiver is configured to detect, at a communication interface, a received signal that suffers from degradations incurred over a communication channel. The receiver applies an adaptive filter to a series of received blocks of a digital representation of the received signal, thereby generating respective filtered blocks, where each received block represents 2N frequency bins, and where N is a positive integer. The receiver calculates coefficients for use by the adaptive filter on a jreceived block as a function of (i) error estimates associated with an (j&#x2212;D&#x2212;1)filtered block, where D is a positive integer representing a number of blocks, and where j is a positive integer greater than (D&#x2212;1); and (ii) an inverse of an approximate covariance matrix associated with the (j&#x2212;D&#x2212;1)received block, where the approximate covariance matrix is a diagonal matrix of size L&#xd7;L, and where L is a positive integer lower than 2N.

Approximation of recursive least squares equalization

A receiver is configured to detect, at a communication interface, a received signal that suffers from degradations incurred over a communication channel. The receiver applies an adaptive filter to a series of received blocks of a digital representation of the received signal, thereby generating respective filtered blocks. The receiver calculates coefficients for use by the adaptive filter on an jreceived block as a function of (i) error estimates associated with an (j&#x2212;D&#x2212;1)filtered block, where D is a positive integer representing a number of blocks, and where j is a positive integer greater than (D&#x2212;1); and (ii) delay compensation terms dependent on an estimate of a difference between coefficients used by the adaptive filter on an (j&#x2212;D&#x2212;1)received block and coefficients used by the adaptive filter on an (j&#x2212;1)received block

Feedback equalization with delay compensation

An optical signal modulated with a stream of symbols comprising a sequence of training symbols is received at a receiver. First equalizer circuitry calculates and applies first coefficients to digital signals representative of the optical signal, thereby resulting in first compensated signals. Second equalizer circuitry calculates second coefficients based on a correlation between the first compensated signals and digital signals representative of the sequence of training symbols and applies the second coefficients to the first compensated signals, thereby resulting in second compensated signals. Third equalizer circuitry calculates and applies third coefficients to the second compensated signals, thereby resulting in third compensated signals. The first, second, and third coefficients compensate for impairments in the optical signal varying at respective first, second, and third rates, where the third rate is higher than the first rate and lower than the second rate.

Training-aided feedforward channel equalization

A method for enhancing a signal directionality in a hearing instrument wherein first and second signals are generated by first and second input transducers that are spaced apart from one another. A first directional signal is derived from the first input signal and the second input signal by applying a second-to-first relative transfer function with respect to a first target angle to the second input signal. The second-to-first relative transfer function is a relative transfer function from the second input transducer to the first input transducer with respect to the first target angle. Similarly, a second directional signal is derived from the second input signal and the first input signal by applying a first-to-second relative transfer function with respect to a second target angle to the first input signal. An angle-enhanced signal is derived from the first directional signal and the second directional signal.

Method for enhancing signal directionality in a hearing instrument

The invention discloses a method for noise reduction in a binaural hearing aid, said binaural hearing aid comprising a first local unit and a second local unit, wherein the method comprises the following steps: generating a first main signal and a first auxiliary signal in the first local unit from an environment sound, and a second main signal in the second local unit from the environment sound, estimating a direction of arrival of a useful sound signal in the environment sound, assigning a first frequency range and a second frequency range, generating a first range beamformer signal in the first frequency range from the first main signal, the first auxiliary signal and the second main signal by imposing at least one spatial condition related to the estimated direction of arrival on the directional characteristic of the first range beamformer signal, generating a second range beamformer signal in the second frequency range from the first main signal and the second main signal by imposing at least one spatial condition related to the estimated direction of arrival on the directional characteristic of the second range beamformer signal, and generating a first local output signal from the first range beamformer signal and the second range beamformer signal, wherein the first local output signal is transduced into a first output sound by a first output transducer of the first local unit.

Method for beamforming in a binaural hearing aid

The invention proposes a method for improving the spatial hearing perception of a binaural hearing aid, said binaural hearing aid comprising a first local unit and a second local unit, wherein in the first local unit, a first input signal is generated from an environment sound by a first input transducer, and a first reference signal is derived from the first input signal, wherein in the second local unit, a second input signal is generated from the environment sound by a second input transducer, and a second reference signal is derived from the second input signal, wherein from the first reference signal and the second reference signal, a first binaural beamformer signal is derived, wherein from the first reference signal and the first binaural beamformer signal, a first coherence parameter is derived, wherein from the first coherence parameter, a first mixing parameter is derived, and wherein the first reference signal and the first binaural beamformer signal are mixed by means of the first mixing parameter in order to generate a first output signal.

Method for improving the spatial hearing perception of a binaural hearing aid

The invention relates to compressing of sparse data sets contains sequences of data values and position information therefor. The position information may be in the form of position indices defining active positions of the data values in a sparse vector of length N. The position information is encoded into the data values by adjusting one or more of the data values within a pre-defined tolerance range, so that a pre-defined mapping function of the data values and their positions is close to a target value. In one embodiment, the mapping function is defined using a sub-set of N filler values which elements are used to fill empty positions in the input sparse data vector. At the decoder, the correct data positions are identified by searching though possible sub-sets of filler values.

Sparse data compression

A method for a multi microphone noise reduction in a complex noisy environment is proposed. A left and a right noise power spectral density for a left and a right noise input frame is estimated for computing a diffuse noise gain. A target speech power spectral density is extracted from the noise input frame. A directional noise gain is calculated from the target speech power spectral density and the noise power spectral density. The noisy input frame is filtered by Kalman filtering method. A Kalman based gain is generated from the Kalman filtered noisy frame and the noise power spectral density. A spectral enhancement gain is computed by combining the diffuse noise gain, the directional noise gain, and the Kalman based gain. The method reduces different combinations of diverse background noise and increases speech intelligibility, while guaranteeing to preserve the interaural cues of the target speech and directional background noises.

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Cantley, Canada

Martin Bouchard