1. "Huge" as in "complicated".
Of course you could sample at 192kHz and have the filter start rolling off at 20kHz and that would make the filter practical again. But then you're already halfway towards an oversampled converter. If the information above 20kHz isn't interesting (you're filtering it away), the ADC is spewing much more data than you're really interested in.
2. When. Why. Whatever. It certainly makes a good reason to do things that way
3. Every other coefficient in a halfband is zero (except for the middle one which is exactly 1) which means you don't need to do any multiplication for those. The length of the filter is not cut in half, but the number crunching is.
4. Money no object I'd still do oversampling. The digital filter would only need to be about 30% more expensive to be perfectly done.
5. Increasing oversampling rate makes analogue filter design easier and converter design harder. So there's going to be an optimum. For Dan's converter topology (discrete ladder), the optimum works out as 2x oversampling.
6. It's a serious question. Converter manufacturers try squeezing out the last few samples worth of latency while DAWs sometimes have hundreds of samples in the FIFO. Who knows one day people will discover the joys of installing a small analogue desk in the live room for the band to make their own headphone mixes on.
In their current converters DAD use only the analogue front-end of the IC and the digital filtering is done separately. The only thing I wasn't sure of is whether they had nonaliasing or halfband. Thanks for updating me.
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Topic: DSD vs PCM - decimation in the ADC? (Read 19409 times)