摘要:
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 phase shifter, 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 phase shifter shifts a phase of the complex-valued subband samples by an arbitrary amount. The high frequency reconstructor modifies at least some of the complex valued subband samples. A phase shifter unshifts a phase of the modified complex-valued subband samples by the arbitrary amount. 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.
摘要:
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 phase shifter, 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 phase shifter shifts a phase of the complex-valued subband samples by an arbitrary amount. The high frequency reconstructor modifies at least some of the complex valued subband samples. A phase shifter unshifts a phase of the modified complex-valued subband samples by the arbitrary amount. 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.
摘要:
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 phase shifter, 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 phase shifter shifts a phase of the complex-valued subband samples by an arbitrary amount. The high frequency reconstructor modifies at least some of the complex valued subband samples. A phase shifter unshifts a phase of the modified complex-valued subband samples by the arbitrary amount. 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.
摘要:
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 phase shifter, 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 phase shifter shifts a phase of the complex-valued subband samples by an arbitrary amount. The high frequency reconstructor modifies at least some of the complex valued subband samples. A phase shifter unshifts a phase of the modified complex-valued subband samples by the arbitrary amount. 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.
摘要:
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.
摘要:
A filter system capable of automatically adjusting bandwidth includes a filter and an adaptive unit. The filter is used for filtering a digital signal to generate an output signal to an application unit. The adaptive unit is used for generating an adjustment signal to the filter according to the digital signal and the output signal. The filter dynamically adjusts bandwidth of the filter according to the adjustment signal.
摘要:
Finite impulse response filters are commonly used in high speed data communications electronics for reducing error rates in multilevel symbol encoding schemes. Schemes such as pulse amplitude modulation and quadrature amplitude modulation may have higher error rates for symbols with low signal to noise ratios. By selectively updating the tap coefficients of the filter based on the symbols received, a more robust, accurate filter can be built.
摘要:
Noise is removed from the digitized output of a sensor, subject to undesired resonance, even when the resonant frequency is unknown or drifts, with sufficiently low phase delay for the sensor to be used in closed-loop control. A very narrow notch filter which removes the resonance-induced noise is recursive (IIR) and therefore has a low phase delay. However, the apparatus which determines the center frequency of the notch filter is non-recursive, and therefore stable. It includes a tunable FIR filter which tracks the same resonance that we wish the IIR filter to remove. Tuning the FIR filter to minimize the output of the FIR filter therefore tunes the notch frequency to align with the resonant frequency. The tuning parameter which adaptively produces this result is suitably scaled and biased, and is applied to the IIR filter.
摘要:
A data processing system (10) implements a symmetrical filtering function about the unity gain line using a single filter (33). The data processing system (10) includes a bus (12), an I/O port (12), memory (16), and a processor (18). The I/O port (12) receives filter control parameters (20) and receives and transmits digitized data. The processor receives the filter control parameters (20) and calculates filter coefficients (44, 48, 52). The processor (18) implements the single stage filter (33) based upon the filter coefficients (44, 48, 52) to filter the digitized input data (22) and produce filtered digitized data (24). The single stage filter (33) may include a band pass, high pass, or low pass function. The single stage filtering function (33) produces a filtering function that is symmetrical about the unity gain line. Variable Q operation of the data processing system (10) allows Q of the filtering function to vary with boost/cut level (34). A method (150), a graphic equalizer (200), and digital telephone (250) also incorporate the symmetrical filtering function.
摘要:
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.