Abstract:
A system and method for representing quasi-periodic (“qp”) waveforms, for example, representing a plurality of limited decompositions of the qp waveform. Each decomposition includes a first and second amplitude value and at least one time value. In some embodiments, each of the decompositions is phase adjusted such that the arithmetic sum of the plurality of limited decompositions reconstructs the qp waveform. Data-structure attributes are created and used to reconstruct the qp waveform. Features of the qp wave are tracked using pattern-ecognition techniques. The fundamental rate of the signal (e.g., heartbeat) can vary widely, for example by a factor of 2-3 or more from the lowest to highest frequency. To get quarter-phase representations of a component (e.g., lowest frequency “rate” component) that varies over time (by a factor of two to three) many overlapping filters use bandpass and overlap parameters that allow tracking the component's frequency version on changing quarter-phase basis.
Abstract:
The document relates to modulated sub-sampled digital filter banks, as well as to methods and systems for the design of such filter banks. In particular, the present document proposes a method and apparatus for the improvement of low delay modulated digital filter banks. The method employs modulation of an asymmetric low-pass prototype filter and a new method for optimizing the coefficients of this filter. Further, a specific design for a 64 channel filter bank using a prototype filter length of 640 coefficients and a system delay of 319 samples is given. The method substantially reduces artifacts due to aliasing emerging from independent modifications of subband signals, for example when using a filter bank as a spectral equalizer. The method is preferably implemented in software, running on a standard PC or a digital signal processor (DSP), but can also be hardcoded on a custom chip. The method offers improvements for various types of digital equalizers, adaptive filters, multiband companders and spectral envelope adjusting filter banks used in high frequency reconstruction (HFR) or parametric stereo systems.
Abstract:
The document relates to modulated sub-sampled digital filter banks, as well as to methods and systems for the design of such filter banks. In particular, the present document proposes a method and apparatus for the improvement of low delay modulated digital filter banks. The method employs modulation of an asymmetric low-pass prototype filter and a new method for optimizing the coefficients of this filter. Further, a specific design for a 64 channel filter bank using a prototype filter length of 640 coefficients and a system delay of 319 samples is given. The method substantially reduces artifacts due to aliasing emerging from independent modifications of subband signals, for example when using a filter bank as a spectral equalizer. The method is preferably implemented in software, running on a standard PC or a digital signal processor (DSP), but can also be hardcoded on a custom chip. The method offers improvements for various types of digital equalizers, adaptive filters, multiband companders and spectral envelope adjusting filter banks used in high frequency reconstruction (HFR) or parametric stereo systems.
Abstract:
A system and method for representing quasi-periodic waveforms, for example, representing a plurality of limited decompositions of the quasi-periodic waveform. Each decomposition includes a first and second amplitude value and at least one time value. In some embodiments, each of the decompositions is phase adjusted such that the arithmetic sum of the plurality of limited decompositions reconstructs the quasi-periodic waveform. Data-structure attributes are created and used to reconstruct the quasi-periodic waveform. Features of the quasi-periodic wave are tracked using pattern-recognition techniques. The fundamental rate of the signal (e.g., heartbeat) can vary widely, for example by a factor of 2-3 or more from the lowest to highest frequency. To get quarter-phase representations of a component (e.g., lowest frequency “rate” component) that varies over time (by a factor of two to three) many overlapping filters use bandpass and overlap parameters that allow tracking the component's frequency version on changing quarter-phase basis.
Abstract:
The present invention relates to nonlinear signal processing, and, in particular, to adaptive nonlinear filtering of real-, complex-, and vector-valued signals utilizing analog Nonlinear Differential Limiters (NDLs), and to adaptive real-time signal conditioning, processing, analysis, quantification, comparison, and control. More generally, this invention relates to methods, processes and apparatus for real-time measuring and analysis of variables, and to generic measurement systems and processes. This invention also relates to methods and corresponding apparatus for measuring which extend to different applications and provide results other than instantaneous values of variables. The invention further relates to post-processing analysis of measured variables and to statistical analysis. The NDL-based filtering method and apparatus enable improvements in the overall properties of electronic devices including, but not limited to, improvements in performance, reduction in size, weight, cost, and power consumption, and, in particular for wireless devices, NDLs enable improvements in spectrum usage efficiency.
Abstract:
Digital signal-processing structure and methodology which feature a time-slice-based digital fabricating engine, and software operating structure operatively associated with that engine structured to operate the engine in a time-slice-based fabrication mode wherein the engine, in a time-differentiated and instantiating manner, functions to fabricate a time-succession of individual, composite wave digital filters. Each of these filters takes the form of (1) a concatenated assembly including one to a plurality of upstream, early-stage, decimate-by-two, signal-processing agencies connected in a cascade series arrangement, with each such agency possessing a first transfer function having a first transition bandwidth, and (2) a single, downstream, later-stage, decimate-by-two, signal-processing agency which possesses a second transfer function having a transition bandwidth which is less than the mentioned first transition bandwidth.
Abstract:
Bit-serial digital filters use numerous flip-flops, which must be reset to a known, all-zero state for testing and start-up purposes. A method for setting a bit-serial digital filter to an all-zero state uses non-resettable flip-flops, which eliminates the increased gate count and current drain overhead of resettable flip-flops. A bit-serial digital filter is constructed using non-resettable flip-flops such as simple non-resettable D flip-flops. When a reset signal is initiated, a reset controller (350) sends an all-zero signal to reset gates (301, 321) positioned at the input to the digital filter and in each feedback loop or unit-delay path. Meanwhile, a bit-serial controller (250) cycles through its control signals to emulate the operation of the bit-serial filter. In two word cycles, each flip-flop in the digital filter will be set to a known, all-zero state, and the all-zero signal is removed to allow normal operation of the filter.
Abstract:
A digital circuit for sampling rate variation and signal filtering includes an input, an output, a lattice wave digital filter having a plurality of filter branches connected to the input, the filter branches each having at least two series-connected filter subgroups with basic filter elements each formed of one two port adaptor made up of adders and multipliers and one time-lag device, a device disposed between the at least two filter subgroups for varying the sampling rate and for generating a phase change in a digital system, and an adder connected between filter branches and the output. A method for constructing the circuit is also provided.
Abstract:
A wave two port circuit arrangement for simulating a resistive elementary port two device for use in a wave digital filter. First and second ports are each arranged for receiving input and output signals. A summing device is connected to the first and second ports and produces a sum signal which represents the sum of the input signals present at the first and second ports. A multiplier is connected to the summing device and to one of the first and second ports for producing a negatively scaled sum signal and feeding it to the one port as part of the output signal present at that port.
Abstract:
In one illustrative embodiment, a filter circuit for discrete signals is provided corresponding in filter function to an analog ladder structure containing inductive and capacitive components. The reactive components of the analog filter structure are realized in the filter circuit as one-port and two-port elements having time delay, while nonreactive components of the analog filter are realized as time-delay-free one-port and plural-port elements. Interface means connect the port elements together in a configuration corresponding to the configuration of the analog ladder structure, with sets of adder and multiplier means of the interface circuitry which couple successive sets of the port elements to each other being correlated with the connective relationship (whether series, series-parallel or parallel) between the corresponding components or branches of the analog ladder structure and with the parameters of the analog components, so that the filter circuit essentially simulates the analog ladder structure in its filter characteristics.