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.

A systolic FIR filter circuit includes a plurality of multipliers, a plurality of sample pre-adders, each respective one of the sample pre-adders connected to a sample input of a respective multiplier, and an output cascade adder chain including a respective output adder connected to a respective multiplier. The output cascade adder chain includes a selectable number of delays between adjacent output adders. An input sample chain has a first leg and a second leg. Each respective one of the sample pre-adders receives a respective input from the first leg and a respective input from the second leg. The input sample chain has, between adjacent sample points in at least one of the legs, a selectable number of sample delays related to the selectable number of output delays. Connections of inputs from the input sample chain to the sample pre-adders are adjusted to account for the selectable number.

The present invention relates to a method and an apparatus for processing a signal, which are used for effectively reproducing an audio signal, and more particularly, to a method and an apparatus for processing a signal, which are used for implementing binaural rendering for reproducing multi-channel or multi-object audio signals in stereo with a low calculation amount. To this end, provided are a method for processing an audio signal including: receiving multi-audio signals including multi-channel or multi-object signals, each of the multi-audio signals including a plurality of subband signals, and the plurality of subband signals including a signal of a first subband group having low frequencies and a signal of a second subband group having high frequencies based on a predetermined frequency band; receiving at least one parameter corresponding to each subband signal of the second subband group, the at least one parameter being extracted from binaural room impulse response (BRIR) subband filter coefficients corresponding to each subband signal of the second subband group; and performing tap-delay line filtering of the subband signal of the second subband group by using the received parameter and an apparatus for processing an audio signal using the same.

The present invention relates broadly to a method of digitally filtering an audio signal by applying a composite audio filter. The composite audio filter may be obtained by applying one audio filter to another audio filter each having the same predetermined sample rate including neighboring sample points. The other audio filter may also include one or more intervening sample points between adjacent of its neighboring sample points. The one audio filter may be applied to the other audio filter at an adjusted sampling rate relative to the other audio filter. The adjusted sampling rate may be inversely proportional to the number of intervening sample points relative to the number of neighboring sample points for the other filter. The frequency response curve for the composite filter derived using the adjusted sampling rate may be more indicative of an idealized lowpass filter. The frequency response with the adjusted sampling rate may display a more bell-shaped characteristic compared with the frequency response without an adjusted sampling rate (shown in broken line detail).

The trans-filter compresses in band AWGN, demodulates input signals and has no threshold due to applied noise. Two frequency selective networks with opposite amplitude vs frequency slopes are designed to remain 180 degrees out of phase over the signal band. Output amplitudes are equal at band center and are summed producing a monotonic amplitude vs frequency characteristic going thru zero at center frequency with abrupt phase reversal. This produces the parabolic output noise density and differentiates applied signals. Absence of nonlinear circuit components and product devices prevents generation of noisenoise products, avoiding the threshold phenomenon. Exponentially modulated digital signals produce output impulses due to the slope and abrupt phase reversal. The impulses have strong fundamental frequency components and may be recovered at baseband without frequency conversion. Cascading trans-filters increases noise reduction and impulse amplitude. The trans-filter algorithm may be used separately or in conjunction with one or more hardware trans-filters.

A control system for controlling operation of an electric appliance is provided. The control system includes a correlator digital filter configured to receive input signals including at least one target signal associated with operation of the electric appliance and extract the target signal from the input signal. The control system also includes a controller operatively coupled with the correlator digital filter and the electric appliance, where the controller is configured to receive the extracted target signal from the correlator digital 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.

An analog-to-digital converter (ADC) using an amplifier-based noise shaping circuit. The amplifier-based noise shaping circuit generates a noise shaping signal. A comparator of the ADC has a first input terminal coupled to an output terminal of a capacitive data acquisition converter that captures an analog input, a second input terminal receiving the noise shaping signal, and an output terminal for observation of the digital representation of the analog input. The amplifier-based noise shaping circuit uses an amplifier to amplify a residual voltage obtained from the capacitive data acquisition converter and provides a switched capacitor network between the amplifier and the comparator for sampling the amplified residual voltage and generating the noise shaping signal.

The present invention relates broadly to a method of digitally filtering an audio signal at a predetermined sample rate by applying a composite audio filter derived at an increase sample rate. The composite audio filter is obtained by combining one audio filter with another audio filter at the increased sample rate. The sample rate of the audio filters may be increased from their predetermined to the increased sample rate using various weighting techniques. The composite filter may provide a frequency response curve with a corner frequency as it approaches the Nyquist frequency whereas the frequency response of a conventional filter (shown in broken line detail) is flat with no effect.

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.