The disclosure relates to a method for operating a selective switching device for signals, a related selective switching device, and a message transmission system having the selective switching device. An embodiment of the method includes the following steps, not necessarily in this order: determining a current temperature in the region of the selective switching device; determining a signal shift of the selective switching device due to the current temperature; adding the signal shift to an input signal of the selective switching device as to receive a compensated signal for which the signal shift due to the current temperature is compensated; and removing the signal shift from an output signal of the selective switching device as to receive a corrected signal for which the compensation is corrected.

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.

An apparatus and method are disclosed for processing an audio signal. The apparatus includes an input interface, a digital filterbank having an analysis part and a synthesis part, a first phase shifter, a spectral envelope adjuster, a second phase shifter, and an output interface. The first phase shifter and the second phase shifter reduce a complexity of the digital filterbank, which includes both analysis and synthesis filters that are complex-exponential modulated versions of a prototype filter.

An apparatus and method are disclosed for filtering an audio signal. The apparatus includes an analysis filter bank, a high frequency reconstructor or parametric stereo processor, and a synthesis filter bank. The analysis filterbank receives real-valued time domain input audio samples and generates complex valued subband samples. The high frequency reconstructor or parametric stereo processor modifies at least some of the complex valued subband samples. The synthesis filter bank receives the modified complex valued subband samples and generates time domain output audio samples. The analysis filter bank comprises analysis filters that are complex exponential modulated versions of a prototype filter with an arbitrary phase shift to reduce a complexity of the filter bank.

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.

An apparatus and method are disclosed for processing an audio signal. The apparatus includes an input interface, a digital filterbank having an analysis part and a synthesis part, a first phase shifter, a spectral envelope adjuster, a second phase shifter, and an output interface. The first phase shifter and the second phase shifter reduce a complexity of the digital filterbank, which includes both analysis and synthesis filters that are complex-exponential modulated versions of a prototype filter.

The document relates to modulated digital filter banks, as well as to methods and systems for the design of such filter banks. In particular, the document discloses a method comprising accessing a time-domain audio signal and applying a first filter bank to the time-domain audio signal, thereby producing a first plurality of subbands of frequency-domain audio data representative of at least a part of the audio signal. The filter bank comprises an decimated modulated filter bank obtained from an asymmetric prototype filter. The method further comprises applying a second filter bank to at least a first subband of the first plurality of subbands of frequency-domain audio data, thereby producing a second plurality of subbands of frequency-domain audio data representative of at least a part of the audio signal. The second modulated filter bank comprises an asymmetric modulated filter bank which includes no decimation. The method further comprises outputting at least the second plurality of subbands of frequency domain audio data.

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.

An apparatus and method are disclosed for filtering an audio signal. The apparatus includes a high frequency reconstructor or parametric stereo processor, a phase shifter, and a synthesis filter bank. The high frequency reconstructor or parametric stereo processor generates modified complex-valued subband samples. A phase shifter unshifts a phase of the modified complex-valued subband samples by an amount. The synthesis filter bank receives the modified complex valued subband samples and generates time domain output audio samples.

An apparatus and method are disclosed for filtering an audio signal. The apparatus includes an analysis filter bank, a phase shifter, a high frequency reconstructor or parametric stereo processor, and a synthesis filter bank. The analysis filterbank receives real-valued time domain input audio samples and generates complex valued subband samples. The phase shifter shifts a phase of the complex-valued subband samples by an amount. The high frequency reconstructor or parametric stereo processor modifies at least some of the complex valued subband samples. A phase shifter then unshifts a phase of the modified complex-valued subband samples by the amount. The synthesis filter bank receives the modified complex valued subband samples and generates time domain output audio samples.