Abstract: A physical layer transceiver for a data channel includes receiver circuitry configured to receive signals on the data channel, transmit circuitry configured to transmit signals onto the data channel, and adaptive filter circuitry coupled to the receiver circuitry and the transmit circuitry and configured to filter the data channel by operating on input frequency-domain data samples to output filtered data samples. The adaptive filter circuitry includes error sample generation circuitry configured to generate error samples representing a difference between a target response and the filtered data samples, arithmetic-only circuitry configured to approximate a windowing function to operate on the error samples, and output sample generation circuitry configured to operate on windowed error samples to provide the output filtered data samples. The comparison circuitry may be configured for time-domain operation and may further be configured to transform the error signals into frequency-domain error signals.

Abstract: A video transform method includes: receiving a quantized transform coefficient for a target block and a transform index for a non-separated secondary transform; inverse-quantizing the quantized transform coefficient; deriving an input transform coefficient size indicating the length of the inverse-quantized transform coefficient to which the non-separated secondary transform has been applied, an output transform coefficient size indicating the length of a modified transform coefficient to which the non-separated secondary transform has been applied, and a transform set mapped to the intra mode of the target block, when the transform index does not indicate that the non-separated secondary transform is not performed; and deriving the modified transform coefficient on the basis of a matrix operation on a transform kernel matrix in the transform set indicated by the transform index, and a transform coefficient list corresponding to the input transform coefficient size.

Abstract: Disclosed is a digital radio-frequency transmitter, which includes a digital logic mixer, a digital power amplifier and an antenna, wherein an output terminal of the digital logic mixer is connected with an input terminal of the digital power amplifier; an output terminal of the digital power amplifier is connected to the antenna; the digital logic mixer is configured to perform logic mixing on baseband data and a radio-frequency local oscillator clock signal which are input into the digital radio-frequency transmitter to generate radio-frequency data; the digital power amplifier is configured to convert the radio-frequency data into an analog power signal; and the antenna is configured to transmit the analog power signal out. According to the digital radio-frequency transmitter of the present application, the circuit layout area and the circuit operation consumption can be effectively reduced.

Abstract: Systems and methods for determining whether a particular feature or change implemented in at least one test store causes a significant change as compared to one or more control stores are discussed. More particularly, techniques for using a time-series clustering algorithm to identify comparable sister stores to a store in which a feature change is being considered are described. Once the sister stores are identified a testing module can perform an A/B testing so as to validate whether a particular feature change being implemented in the test store causes a significant change as compared to the control stores.

Abstract: An image decoding method according to the present document comprises the steps of: receiving a quantized transform coefficient for a target block and a transform index for non-separable quadratic transform; deriving transform coefficients by inversely quantizing the quantized transform coefficient; deriving corrected transform coefficients on the basis of a transform kernel matrix in a predetermined transform set indicated by the transform index; and deriving residual samples for the target block on the basis of inverse linear transform for the corrected transform coefficients, wherein when the target block is divided into a predetermined number of sub-blocks and is coded by intra prediction, the corrected transform coefficients are derived in units of the sub-blocks, and the transform index is received for the target block.

Abstract: There is provided mechanisms for obtaining channel conditions per antenna element of an antenna array comprising N antenna elements. The antenna array is configured for communications using orthogonal sub-carriers. A method is performed by a radio transceiver device. The method comprises obtaining channel conditions of a radio propagation channel per sub-carrier for signals received on the sub-carriers by the antenna array. A set of receiving entities are associated with the sub-carriers in an interleaved manner such that each receiving entity is interleaved over a respective disjoint subset of sub-carriers. The method comprises transforming the channel conditions at least for those sub-carriers located within a coherence bandwidth of the radio propagation channel to channel conditions for the N antenna elements by using a relation that maps the receiving entities to the N antenna elements.

Abstract: A broadband linear processing system includes a pre-processing module and a set of M linear processors coupled to the pre-processing module, M being an integer greater than 1. The pre-processing module includes a wavefront multiplexer having M input ports and M output ports. The wavefront multiplexer receives M input signals at the M input ports, performs a wavefront multiplexing transform on the M input signals and outputs M narrowband signal streams at the M output ports. The wavefront multiplexing transform has an inverse. Each of the M linear processors receives and processes a corresponding one of the M narrowband signal streams, and outputs a corresponding one of M processed narrowband signal streams.

Abstract: A broadband digital beam forming system comprises a set of Q pre-processing modules, Q being an integer greater than or equal to 2, and a set of M digital beam forming modules in communication with the Q preprocessing modules. Each of the Q preprocessing modules receives a respective one of Q broadband input signal streams and outputs M narrowband signal streams, M being an integer greater than or equal to 2. The total number of narrowband signal streams outputted by the Q pre-processing modules is Q times M. Each of the M digital beam forming modules receives corresponding Q narrowband signal streams of the Q times M narrowband signal streams, and outputs R beam signals, R being an integer greater than or equal to 1. The system further comprises a set of R post-processing modules in communication with the M digital beam forming modules.

Abstract: A radar sensing system for a vehicle includes a transmitter configured for installation and use on a vehicle and able to transmit radio signals. The radar sensing system also includes a receiver and a processor. The receiver is configured for installation and use on the vehicle and able to receive radio signals. The received radio signals include transmitted radio signals that are reflected from objects in the environment. The received radio signals further include radio signals transmitted by at least one other radar system. The processor samples the received radio signals to produce a sampled stream. The processor is configured to control an adaptive filter. Responsive to the processor, the adaptive filter is configured to filter the sampled stream, such that the radio signals transmitted by the at least one other radar system are removed from the received radio signals.

Abstract: Methods and systems for amplitude modulation in a parametric speaker system are provided that perform truncated double sideband (TDSB) frequency modulation of audio signal in which most of the processing is performed in the frequency domain, thus permitting use of fast processing techniques for amplitude modulation (AM) and filtering and reducing computation cost over time domain processing. A maximum envelope value of the time domain audio signal may be to the carrier signal in the frequency domain that avoids emitting the carrier signal when the input signal level is low or mute. The application of the envelope value may be smoothed to reduce discontinuity at input block boundaries.

Abstract: New hybrid filters are presented based on time and transform domain structures. The hybrid filters have a combined benefit from the advantages obtained by the time and transform domain structures. The overall efficiencies are drawn from combining the pre- and post-processing of the time domain and block based transform domain structures. Further improvements are obtained by interchanging block construction and transforms with linear operations in the pre- and post-processors. The hybrid structures apply to single input, single output, multiple input, and multiple output structures. For the multi input and multi output structures further improvements are obtained by having common processing blocks for the input(s) and common processing blocks for the output(s). They hybrid filters are also efficient in topologies where filter outputs are combined via linear operation(s) generating combined results.

Abstract: The present invention relates to a method for processing a digital input signal by a Finite Impulse Response, FIR, filtering means, comprising partitioning the digital input signal at least partly in the time domain to obtain at least two partitions of the digital input signal; partitioning the FIR filtering means in the time domain to obtain at least two partitions of the FIR filtering means; Fourier transforming each of the at least two partitions of the digital input signal to obtain Fourier transformed signal partitions; Fourier transforming each of the at least two partitions of the FIR filtering means to obtain Fourier transformed filter partitions; performing a convolution of the Fourier transformed signal partitions and the corresponding Fourier transformed filter partitions to obtain spectral partitions; combining the spectral partitions to obtain a total spectrum; and inverse Fourier transforming the total spectrum to obtain a digital output signal.

Abstract: A rectangular wave for determining a range of a weight function is transformed to frequency domain by an FFT or the like, and after being multiplied by a window function (BlackmanHarris window function, for example) generated on a frequency axis by a multiplier, the frequency domain is transformed again to the time domain by an IFFT or the like thereby to generate a weight function.

Abstract: An apparatus for performing improved audio processing may include a processor. The processor may be configured to divide respective signals of each channel of a multi-channel audio input signal into one or more spectral bands corresponding to respective analysis frames, select a leading channel from among channels of the multi-channel audio input signal for at least one spectral band, determine a time shift value for at least one spectral band of at least one channel, and time align the channels based at least in part on the time shift value.

Abstract: A method for calculating coefficients of a filter and a method for filtering are provided. The invention directly factorizes a specific function in a cepstrum domain by spectral factorization and cepstrum technique to obtain coefficients of denominator function from the filter. In other words, the invention adopts a non-iterative algorithm to reduce computational complexity and avoid convergence due to calculating coefficients. Besides, the specific function of the invention includes a compensation function, so that a Fourier transform with greatly reduced size can be utilized in the spectral factorization to greatly save the computations and keep good system performance at a receiver.

Abstract: A method and apparatus are disclosed for use with multiple input, multiple output (MIMO) signal processing techniques, which reduce the amount of memory and memory bandwidth used to store and access filter coefficients by compressing a filter coefficient based at least in part on one or more neighboring filter coefficients for storage and decompressing the filter coefficients when retrieved. The decompressed filter coefficients can be used with a MIMO filtering technique, and/or can be used to compress or decompress additional coefficients.

Abstract: In signal processing using digital filtering the representation of a filter is adapted depending on the filter characteristics If e.g. a digital filter is represented by filter coefficients for transform bands Nos. 0, 1, . . . , K in the frequency domain, a reduced digital filter having coefficients for combined transform bands, i.e. subsets of the transformed bands, Nos. 0, 1, . . . , L, is formed and only these coefficients are stored. When the actual filtering in the digital filter is to be performed, an actual digital filter is obtained by expanding the coefficients of the reduced digital filter according to a mapping table and then used instead of the original digital filter.

Abstract: We disclose an optimal hardware implementation of the FFT/IFFT operation that minimizes the number of clock cycles required to compute the FFT/IFFT while at the same time minimizing the number of complex multipliers needed. An input module combines a plurality of inputs after applying a multiplication factor to each of the inputs. At least one multiplicand generator generates multiplicands. At least two complex multiplier modules perform complex multiplications with at least one of the complex multiplier modules receiving an output from the input module. A map module receives outputs of the at least two complex multiplier modules, the map module selecting and applying a multiplication factor to each of the outputs received to generate multiple outputs. Finally, an accumulation module receives and performs an accumulation task on each of the multiple outputs of the map module thereby generating a corresponding number of multiple outputs.

Abstract: The present invention relates to a digital data filtering circuit. This digital data filtering circuit is able to implement the calculation steps of a discrete transform of a set of 8 original data (w), and calculating an inverse discrete transform of the set of transformed data thus obtained. For this purpose, it comprises a first filtering module (FILo1) intended to filter the odd transformed data or the 3 odd transformed data items having the highest frequencies in the set of transformed data, and a second filtering module (FILo2) connected to the first filtering module and intended to filter the 2 odd transformed data having the highest frequencies in the set of transformed data.

Abstract: A non-invasive motion and respiration monitor receives impulses from a subject's movement, heartbeat, and respiration. The raw signal is biased and digitized, and a signal processor applies a Fast Fourier Transform to the signal. The transformed signal is filtered to isolate the component representing heart rate from the component representing respiration. An Inverse Fast Fourier Transform is then applied to the component signals, which are sent to a processor. The processor is programmed to detect irregularities in the respiration and heart rate. If severe irregularities or complete cessation is detected in either signal, a mechanical stimulator is actuated to try to stimulate the subject, and an alarm is sounded to alert a caregiver such as a parent or nurse.

Abstract: A method for data processing includes transforming measurement data acquired in the time domain during an oilfield operation into a second domain to produce transformed data; identifying distortions in the transformed data; removing the distortions from the transformed data; and transforming back from the second domain to the time domain to produce cleaned-up data. The transforming measurement data may use a Fourier transform or a wavelet transform. The method may further include compressing the cleaned-up data or reconstructing signals from the cleaned-up data. A method for data processing includes decomposing measurement data, which are acquired in an oilfield operation, using a low pass filter to produce a first dataset; decomposing the measurement data using a high pass filter to produce a second dataset; removing distortions from the second dataset to yield a corrected second dataset; and reconstructing a corrected dataset from the first dataset and the corrected second dataset.

Abstract: A method and apparatus adapted to calibrate a signal path of a signal analysis system such that loading effects of the system are substantially removed from measurements of a device under test. A signal under test from the device under test is coupled to a test probe in the signal path and used with selectable impedance loads in the test probe to characterize transfer parameters of the device under test. An equalization filter in either the frequency or time domain is computed from the device under test transfer parameters for reducing in signal error attributable to the measurement loading of the device under test.

Abstract: Disclosed is a method of providing signal processing operations, including convolution/filtering, where each parallel stream is processing signals at a lower rate than the signal data rate itself while still resulting in an overall signal processing rate suitable for high rate signal processing. This allows devices with a lower signal processing rate to be used in the parallel processing paths, avoiding many of the problems facing prior art high data rate signal processing. The invention converts a signal from the time domain to the frequency domain and takes advantage of associative and communicative properties of data processing to perform signal processing operations in parallel.

Abstract: One embodiment of the present invention provides a system that reconstructs a high-resolution signal from a set of low-resolution quantized samples. During operation, the system receives a time series containing low-resolution quantized signal values which are sampled from the high-resolution signal. Next, the system performs a spectral analysis on the time series to obtain a frequency series for the low-resolution quantized signal values. The system next selects a subset of frequency terms from the frequency series which have the largest amplitudes. The system then reconstructs the high-resolution signal by performing an inverse spectral analysis on the subset of the frequency terms.

Abstract: The present invention relates generally to the problem of filtering, decimation or interpolation and frequency conversion in the digital domain, and more particularly to its use in wideband multichannel receiver, channelization, and transmitter, de-channelization, structures. The invention combines a stand-alone fast convolution algorithm which is further modified and then combined with additional signal processing. By intelligently splitting the filtering effort between the modified fast convolution algorithm block and an additional signal processing block a synergy is created between the two blocks which provides for decreased costs, reduced delay and a reduction in the size of the Fast Fourier Transforms (FFTs). The resulting advantages are especially useful in any system handling multiple channels simultaneously, but especially where there exist strict requirements on both delay and on input Fast Fourier Transform (FFT) size.

Abstract: A complex multiplier is described for filtering a signal in the frequency domain. In one embodiment, additional, independent frequency coefficients are supplied by the complex multiplier so that gain and/or phase of the signal may be independently modified (i.e., gain may be modified without affecting phase and vice-versa).

Abstract: A method and apparatus for real-time derivation of precise digital clock edges and synchronous logic samples from a digital signal having a clock channel and at least one data channel acquires a plurality of temporally offset analog samples during each of a sequence of sample periods and from consecutive samples where there is a logic level transition estimates an edge time. From the edge times for the clock channel an offset is added and applied to the at least one data channel to determine the synchronous logic samples for the data channel at each offset clock edge time.

Abstract: A matched-phase noise filter includes an analog to digital converter (ADC) for receiving an analog composite, noise-dominated signal containing a signal of interest and producing a digital composite signal, an input/output port receiving the digital composite signal and providing a matched-phase signal, and a processor receiving the digital composite signal via the I/O port and generating the matched-phase signal. According to one aspect of the invention, the signal to noise ratio between the signal of interest and a noise component within the digital composite signal is increased by approximately an order of magnitude, based on an actual spectrum of the digital composite signal and an estimated spectrum of the noise component, and independent of the particular form of the signal of interest. A method of matched-phase noise filtering for improving the SNR of a noise dominated signal independent of the form of the signal is also described.

Abstract: A system (100) for signal processing using one or more filters includes a controller (102) that partitions an impulse response of a filter into a plurality of impulse response blocks and calculates a Discrete Fourier Transform (DFT) of each impulse response block using a Fast Fourier Transform (FFT) algorithm. A processor (104) that is coupled to the controller (102) receives an input sample block including samples of a signal to be processed and receives the DFT of each impulse response block from the controller (102). The processor (104) calculates a DFT of the input sample block using an FFT algorithm, performs a spectral multiplication of the DFT of the input sample block with the DFT of each impulse response block, overlap-adds the blocks resulting from each spectral multiplication to create an output spectral block, performs an inverse FFT on the output spectral block to create an output sample block, and communicates the output sample block.

Abstract: The present invention is directed to a method of providing signal processing operations, including convolution/filtering, where each parallel stream is processing signals at a lower rate than the signal data rate itself while still resulting in an overall signal processing rate suitable for high rate signal processing. This allows devices with a lower signal processing rate to be used in the parallel processing paths, avoiding many of the problems facing prior art high data rate signal processing. The invention converts a signal from the time domain to the frequency domain and takes advantage of associative and communicative properties of data processing to perform signal processing operations in parallel.

Abstract: A signal processing algorithm has been developed in which a filter function is extracted from degraded data through mathematical operations. The filter function can then be used to restore much of the degraded content of the data through use of any deconvolution algorithm. This process can be performed without prior knowledge of the detection system, a technique known as blind deconvolution. The extraction process, designated Self-deconvolving Data Reconstruction Algorithm (SeDDaRA), has been used successfully to restore digitized photographs, digitized acoustic waveforms, and other forms of data. The process is non-iterative, computationally efficient, and requires little user input. Implementation is straight-forward, allowing inclusion into all types of signal processing software and hardware.

Abstract: The present invention relates generally to the problem of filtering, decimation or interpolation and frequency conversion in the digital domain, and more particularly to the use of the stand-alone or improved modified fast convolution algorithm in wideband multichannel receiver, channelization, and transmitter, de-channelization, structures of a radio communication system. The invention consists of essentially 3 steps: making sure that we use different overlaps on consecutive blocks that, on average, give the same overlap on both the input and output ends; aligning the signal in consecutive blocks of time; and compensating for phase shifts due to frequency shifting. The essence of the invention is that it decouples the input and output transform lengths in the fast convolution algorithm from each other and from the overlap, making it possible to use any transform length on the input together with any transform length on the output and at the same time use any overlap.

Abstract: A moving averaging filter which does not propagate a calculation error is provided reducing the size of the hardware. This moving average filter has a data holding unit for holding multiple successive data, a coefficient storing unit for storing coefficients, a first adder which calculates the sum of a pair of data of a prescribed combination held in the data holding unit, a multiplier which multiplies the sum to coefficient data obtained from the coefficient storing unit, and a second adder which adds up-a prescribed number of multiplication results produced by the multiplier.

Abstract: An implicit DST-based filter having characteristics defined by a linear convolution kernel that may be causal or noncausal-symmetric. The filter filters an information signal composed of blocks of discrete cosine transform (DCT) coefficients to generate a filtered information signal also composed of blocks of DCT coefficients. The filter comprises multiplying matrices, a deriving module, matrix multiplying modules and a summing module. The multiplying matrices are obtained by absorbing a cosine-to-sine transform and a sine-to-cosine transform into kernel matrices derived from the linear convolution kernel. The deriving module derives intermediate blocks of DCT coefficients from neighboring ones of the blocks of DCT coefficients constituting the information signal. The matrix multiplying modules multiply the intermediate blocks of DCT coefficients by the multiplying matrices.

Abstract: A platform-independent software-implemented digital signal processing system in a computer system. The digital signal processing system is comprised of a digital signal processing unit implemented in C program code which receives data and applies a process to generate therefrom a computer file representative of a processed response. The digital signal processing system also is comprised of a graphical user interface implemented in Java program code, which generates two-dimensional graphic images and text on the computer system's user output device, and is for receiving user-selected process parameters and for displaying the results of the processed response. In addition, the digital signal processing system is comprised of an interface code which provides communication between the graphical user interface and the digital signal processing unit.

Abstract: In a method is disclosed for transmitting and receiving a transmission data signal, after the transmission data signal is generated, it is modulated by a first time spread root Nyquist filter to generate a transmission signal. The signal is then transmitted and received, whereupon the signal is transformed into a baseband signal. The baseband signal is then demodulated through a second time spread root Nyquist filter to revive the transmission data signal. An apparatus is also disclosed that performs this method. In addition, methods of separately transmitting and receiving signals are also described.

Abstract: A signal processing system for generating a signal component reduced output signal by estimating the power spectrum of a non-stationary signal component within a non-stationary input signal is disclosed. The estimated power spectrum of the signal component is determined by using a gamma delay line (GDL) filter. The weights of the GDL filter are adaptively updated in accordance with a detection method that is designed to detect the presence of the non-stationary signal component. By subtracting the estimated power spectrum of the signal component from the power spectrum of the entire input signal, a signal component reduced output signal can be generated.