Abstract:
An adaptive, recursive digital wave filter includes a transfer filter and a gradient filter each having a respective adaptor with a given coefficient. A method and circuit configuration for avoiding overflows in the filter include multiplying a signal emitted by the transfer filter to the gradient filter by a scaling factor being equal to the difference between 1 and the amount of the given coefficient.
Abstract:
A digital signal processing arrangement including a wave digital filter having quantizing means for limiting the signal waves present to a predetermined word length by means of controlled rounding. "Controlled rounding" means that a limit-cycle-free output signal is obtained when a constant input signal is present. If the outgoing signal waves from the N-port adaptor are equal to b.sub.o (j,k), and the signal waves applied to the adaptor are equal to a(j,k), b(j,k) is equal the desired and quantized value of b.sub.o (j,k), where j represents the number of the port, then the process of controlled rounding quantizes b.sub.o (j,k) in such a way that ##EQU1## Here the c(i,j) represent constant coefficients, and r the number of independent information signal waves.
Abstract:
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.
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:
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.
Abstract:
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.
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:
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.
Abstract:
An apparatus and method are disclosed for filtering an audio signal are disclosed. The apparatus includes an analysis filter bank, a high frequency reconstructor or a 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.
Abstract:
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.