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.

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.

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.

A method for data filtering includes segmenting a to-be-detected vector to obtain k to-be-detected sub-vectors, respectively performing an inner product operation on the k to-be-detected sub-vectors and corresponding detection vectors among preset k detection vectors to obtain k first operation results, determining a first operation result whose value is the maximum among the k first operation results and obtaining an identifier of a detection vector corresponding to the first operation result, where a detection vector is in a one-to-one correspondence to an identifier, and mapping the to-be-detected vector to a preset data filter according to the obtained identifier of the detection vector corresponding to the first operation result whose value is the maximum, and determining, using the data filter, whether to filter out the to-be-detected vector.

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.

Typically, complex systems require a separate and expensive equalizer at the output of an analog-to-digital converter (ADC). Rather than providing a separate equalizer, the effective Signal Transfer Function (STF) of a Multi-stAge noise SHaping (MASH) ADC can be modified by leveraging available digital filtering hardware necessary for quantization noise cancellation. The modification can involves adding calculations in the software previously provided for computing digital quantization noise cancellation filter coefficients, where the calculations are added to take into account equalization as well. As a result, the signal transfer function can be modified to meet ADC or system-level signal-chain specifications without additional equalization hardware. The method is especially attractive for high-speed applications where magnitude and phase responses are more challenging to meet.

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.