The behavior of a NOS DAC and an analog filter may be emulated by electronic components of an integrated circuit (IC) by upsampling data and applying a digital filter to the upsampled data. For example, the IC may include a zero-order-hold circuit that upsamples data from a first input sample rate to a second, higher input rate. The upsampled data may be passed to an Asynchronous Sample Rate Converter (ASRC) that performs further upsampling (e.g., from 8*Fs-64*Fs). The upsampled data may be passed to a digital low pass filter. The digital low pass filter may emulate, for example, a response of a fifth order Butterworth analog filter to mimic the effect of analog processing. The IC may integrate the upsampling circuit, the low pass digital filter, a digital-to-analog converter (DAC) and an amplifier to provide an audio solution for playing high-fidelity music in a mobile device.

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.

A method of digitally filtering an audio signal using an adjusted audio filter. The adjusted audio filter is represented by an impulse response including a waveform in its time domain represented by a sine function of absolute values. A composite audio filter is derived from two adjusted audio filters although any number of filters may be used. The composite audio filter generally includes a bank of the filters which together define a frequency bandwidth representative of the audio signal or spectrum to be filtered. Also a bandpass filter is constructed by combining frequency responses for sine components of absolute values integrated from 0 to bpf and sine components of absolute values integrated from 1/bpf to 0. The frequency response may be the sum of the frequency responses for each of the filters used to create the composite bandpass filter.

A discrete time (DT) lowpass filter having various advantages is described. In an exemplary design, the DT lowpass filter includes a decimating DT filter (which may include a passive DT FIR filter and/or a passive DT IIR filter) and an active DT filter. The decimating DT filter receives a first DT signal at a first sample rate, filters and decimates the first DT signal by a factor of N, and provides a second DT signal at a second sample rate lower than the first sample rate. N may be greater than one. The active DT filter filters the second DT signal and provides a third DT signal at the second sample rate. A sampler samples a continuous time signal and provides the first DT signal. The sampler may further double the voltage of the first DT signal relative to the voltage of the continuous time signal.