Abstract: Examples described herein include systems and methods which include wireless devices and systems with examples of mixing input data with coefficient data specific to a processing mode selection. For example, a computing system with processing units may mix the input data for a transmission in a radio frequency (RF) wireless domain with the coefficient data to generate output data that is representative of the transmission being processed according to a specific processing mode selection. The processing mode selection may include a single processing mode, a multi-processing mode, or a full processing mode. The processing mode selection may be associated with an aspect of a wireless protocol. Examples of systems and methods described herein may facilitate the processing of data for 5G wireless communications in a power-efficient and time-efficient manner.

Abstract: A system for generating stereo-channel audio signals with surround information is disclosed. The system includes a surround mapping unit configured to receive signals from a number of audio channels and generate a pair of stereo-channel audio signals based on the audio channels. The pair of stereo-channel audio signals includes binaural and spatial information. The system also includes a stereo-channel encoder configured to receive and encode the pair of stereo-channel audio signals from the surround mapping unit thereby generating a pair of encoded stereo-channel audio signals. The system further includes a stereo-channel decoder configured to receive and decode the pair of encoded stereo-channel audio signals thereby obtaining the pair of stereo-channel audio signals. The pair of stereo-channel audio signals are capable of being used to generate surround effect.

Abstract: An image processing method and system using a low-complexity scheme is provided. According to one aspect of the method, input components from a RGB model are used directly to calculate the down-sampled components of a YCbCr model. In an exemplary instance where average down-sampling and a down-sampling rate of “4:2:0” are used, the following equations are used to derive the down-sampled components of the YCbCr model: Y l ? = ? 0.29900 ? R i + 0.58700 ? G l + 0.11400 ? B l Cb ? = ? - 0.04219 ? ? i = 0 3 ? ? R l - 0.082815 ? ? i = 0 3 ? ? G l + 0.12500 ? ? i = 0 3 ? ? B l + 2 SP / 2 Cr ? = ? 0.12500 ? ? i = 0 3 ? ? R i - 0.10467 ? ? i = 0 3 ? ? G i - 0.02033 ? ? i = 0 3 ? ? B i + 2 SP / 2 where Ri, Gi and Bi are three input components of the color conversion for the pixel i and “SP” represents a specified sample precision under the RGB model.

Abstract: An improved audio compression scheme is provided. The scheme uses an excitation pattern to more efficiently provide audio signal compression. Under the scheme, an input signal is transformed to the frequency domain. Next, the excitation pattern corresponding to the transformed input signal is calculated. Bit allocation processing is then performed based on the excitation pattern. Frequencies are then coded based on the results of the bit allocation processing. Finally, bitstream packing is performed to generate the output coded audio bit stream. In one exemplary implementation, the audio compression scheme is implemented in an encoder.

Abstract: The invention provides a DMT or OFDM equalization system that deals with each tone not only separately, but also globally, which provides better overall performance. In the inventive system, only M+T variables are required to be determined, where M is the number of tones and T represents the number of SIRF taps. The invention defines a matrix R, in which optimal coefficients are found as the eigenvector corresponding to the smallest eigenvalue of the matrix R. The dynamic range of all variables is limited in a system in accordance with the present invention, which provides ease of hardware implementation. Furthermore, the inventive system retains the advantages of per-tone equalization by providing a smoother signal-to-noise ratio (SNR) distribution function versus synchronization delay. In addition, no effort is wasted on the equalization of unused tones, because it is unnecessary to determine the coefficients for unused tones.

Abstract: A system for frequency-domain scaling for DCT computation. Scale factors are applied to coefficients during the final steps of composition of 2-point DCTs. The number of multiplications and required precision are reduced. Fixed values for various scale factors can be computed and stored prior to executing the DCT so that performance can be improved. The fixed values are derived by knowing the length of the time-domain sequence. Some fixed values can be derived independently of the length of the time-domain sequence. The approach of the invention can also reduce the number of multiplications to compute the transform, and allow smaller bit-width sizes by reducing the number of required high-precision calculations.

Abstract: The invention provides a DMT or OFDM equalization system that deals with each tone not only separately, but also globally, which provides better overall performance. In the inventive system, only M+T variables are required to be determined, where M is the number of tones and T represents the number of SIRF taps. The invention defines a matrix R, in which optimal coefficients are found as the eigenvector corresponding to the smallest eigenvalue of the matrix R. The dynamic range of all variables is limited in a system in accordance with the present invention, which provides ease of hardware implementation. Furthermore, the inventive system retains the advantages of per-tone equalization by providing a smoother signal-to-noise ratio (SNR) distribution function versus synchronization delay. In addition, no effort is wasted on the equalization of unused tones, because it is unnecessary to determine the coefficients for unused tones.

Abstract: An adaptive binaural beamforming system is provided which can be used, for example, in a hearing aid. The system uses more than two input signals, and preferably four input signals. The signals can be provided, for example, by two microphone pairs, one pair of microphones located in a user's left ear and the second pair of microphones located in the user's right ear. The system is preferably arranged such that each pair of microphones utilizes an end-fire configuration with the two pairs of microphones being combined in a broadside configuration. Signal processing is divided into two stages. In the first stage, the outputs from the two microphone pairs are processed utilizing an end-fire array processing scheme, this stage providing the benefits of spatial processing. In the second stage, the outputs from the two end-fire arrays are processed utilizing a broadside configuration, this stage providing further spatial processing benefits along with the benefits of binaural processing.

Abstract: A computing machine capable of performing multiple operations using a universal computing unit is provided. The universal computing unit maps an input signal to an output signal. The mapping is initiated using an instruction that includes the input signal, a weight matrix, and an activation function. Using the instruction, the universal computing unit may perform multiple operations using the same hardware configuration. The computation that is performed by the universal computing unit is determined by the weight matrix and activation function used. Accordingly, the universal computing unit does not require any programming to perform a type of computing operation because the type of operation is determined by the parameters of the instruction, specifically, the weight matrix and the activation function.

Abstract: A system for frequency-domain scaling for DCT computation. Scale factors are applied to coefficients during the final steps of composition of 2-point DCTs. The number of multiplications and required precision are reduced. Fixed values for various scale factors can be computed and stored prior to executing the DCT so that performance can be improved. The fixed values are derived by knowing the length of the time-domain sequence. Some fixed values can be derived independently of the length of the time-domain sequence. The approach of the invention can also reduce the number of multiplications to compute the transform, and allow smaller bit-width sizes by reducing the number of required high-precision calculations.

Abstract: A feedback cancellation system for a hearing instrument is described. The feedback cancellation system includes both a constrained adaptive filter, the constrained adaptive filter can be constrained by initiation coefficients, as well as an adaptive gain modification. The adaptive gain modification allows the feedback cancellation to respond to substantially different environments from the test environment used to derive the initiation coefficients for the constrained adaptive filter.

Abstract: The present invention comprises an adaptive directionality dual microphone system in which the time domain data from the first and second microphones is converted into frequency domain data. The frequency domain data is then manipulated to produce a noise-canceled signal which is converted in an Inverse Fourier Transform block into noise-cancel time domain data.

Abstract: A computing machine capable of performing multiple operations using a universal computing unit is provided. The universal computing unit maps an input signal to an output signal. The mapping is initiated using an instruction that includes the input signal, a weight matrix, and an activation function. Using the instruction, the universal computing unit may perform multiple operations using the same hardware configuration. The computation that is performed by the universal computing unit is determined by the weight matrix and activation function used. Accordingly, the universal computing unit does not require any programming to perform a type of computing operation because the type of operation is determined by the parameters of the instruction, specifically, the weight matrix and the activation function.

Abstract: An adaptive binaural beamforming system is provided which can be used, for example, in a hearing aid. The system uses more than two input signals, and preferably four input signals. The signals can be provided, for example, by two microphone pairs, one pair of microphones located in a user's left ear and the second pair of microphones located in the user's right ear. The system is preferably arranged such that each pair of microphones utilizes an end-fire configuration with the two pairs of microphones being combined in a broadside configuration. Signal processing is divided into two stages. In the first stage, the outputs from the two microphone pairs are processed utilizing an end-fire array processing scheme, this stage providing the benefits of spatial processing. In the second stage, the outputs from the two end-fire arrays are processed utilizing a broadside configuration, this stage providing further spatial processing benefits along with the benefits of binaural processing.