I've really enjoyed your videos, many thanks for your work producing them. In keeping with your message in the video about breaking things I was curious to see what would happen as you pushed the signal up to and beyond the Nyquist frequency. Would the filters mean that it would simply fade out smoothly (what order filter is employed?)? I suppose I could check this out myself, but would be interested to hear your answer. Also is the Gibbs phenomenon audible to any degree? --[[User:Stuarticus|Stuarticus]] 14:00, 11 June 2013 (PDT)
: If the filter is smooth, it will fade smoothly. If the filter is sharp, it will drop off suddenly. Some DACs put the transition band of the anti-imaging filters straddling the Nyquist frequency or slightly past, so you might even see the signal fold back about Nyquist as it drops off (this used to be more common about 10 years ago). So long as no significant aliasing reaches back under 20kHz, this is just one of many arbitrary design decisions that don't really affect the audio quality. The order of the digital filter used in the upsampling stage can be nearly anything, but they're not likely to be as huge as many software resampling filters, where a linear-phase FIR of 512 or 1024 taps is common. The analog filter stages, if they're there at all, are unlikely to be more than a handful of taps. Detailed spec sheets should say outright, and if they don't they should at least usually mention the approximate slope. --[[User:Xiphmont|Xiphmont]] 03:45, 15 June 2013 (
I just watch the video and has one question. During demonstration which Monty has feed generator at various frequency and convert to digital at 44.1KHz and convert back to analog and show the analog result at second oscilloscope which is the good way to show the whole digital way of encode and decode. At 1KHz sine wave input, when sampling at 44.1KHz, there will be 44.1 sampling per sine wave or 22 sampling per half sine wave which is not too bad to represent the sine wave and the output from digital-to-analog should still be quite close to original sine wave. However, at input of 20KHz, there will be only 2.2 sampling per sine wave or just 1 sampling per half wave and it merely enough to represent sine wave. My question is if there is just one sampling per half wave, how can the output from digital-to-analog is still be very good sine wave. If 20KHz is sampling at 192KHz, there will be 9.6 sampling per sine wave or 4.8 sampling per half wave which is far from perfect but still better than one. Does this mean that we should have better output if we increasing the sampling from 44.1KHz to 192KHz?