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.

To suppress an unstable operation of a device caused by filtering, an output selection unit is configured to acquire an unfiltered value representing an input signal (signal before filtering) stored in an input holding unit, a filtered value representing a signal acquired by filtering the input signal by a filtering unit, and a previous output value representing an output signal output at a previous time, select a middle value out of the filtered value, the unfiltered value, and the previous output value, and set the selected middle value as a current output value of an output signal.

An apparatus and method are disclosed for filtering and performing high frequency reconstruction of an audio signal. The apparatus includes an analysis filter bank, a high frequency reconstructor, 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 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.

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.

An Analysis Polyphase Filter (APPF) for shifting a selected passband of an input signal to a passband signal, having a Q=M.Math.B/A decimation factor, B and A being co-primes, the APPF comprising M.Math.B Paths. Each path includes (i) M.Math.B/A multiplication Sub-paths, and (ii) a Sub-path Fusion Circuit, which is configured to generate fused multiplication products responsively to a sum of the multiplication n products generated by each of the multiplication sub-paths.

An apparatus for calibrating a digital filter to replicate a transfer function of a signal processing device comprising a detector with a first input to receive a first signal; a second input to receive a response signal of the signal processing device to the first signal; a signal modification block; a comparison block and a decimation block; wherein the comparison-block compares a phase and amplitude of the first signal after a correction has been applied by the signal modification block and decimation has been applied by the decimation block; and a feedback loop; wherein detector is configured to determine at least a feedback control signal at a first frequency and a second frequency, different to the first frequency and determine calibration information for programming of the transfer function of said digital filter.