Even a 8192 samples long filter doesn't cause anything I could hear. Maybe because the passband has to be a lower to make the filter audible, e.g. 18 kHz instead of 21 kHz? Anyway the filter is quite steep. -6 dB at 21 kHz and -90 dB at 21.0157 kHz.
I recall a study in which members of an orchestra were asked to compare two tones very close in frequency (I think it was A440).Many of the members were able to distinguish tones only 1 Hz apart, and one member was able to distinguish tones only 0.1 Hz apart.I then ask myself, did he need to hear the tones for a full ten seconds each so that there was one full cycle difference between them?
You want to make the filter as steep as possible, not with as much rejection as possible. If I had access to matlab right now I'd give you firpm paramaters, but I tried what I recall in octave via remez and it just died instead.
Regarding what happens at the edges, sinc pulses die out fairly rapidly, don't they; especially when realistic levels are taken into account?
sox castanets-2_2496.wav output.wav sinc -a 90 -n 32767 -21k
A simple impulse will set them in motion, just like any other mechanical device.
For 90dB attenuation low-pass, 32767 taps, 6dB corner @ 21kHz; which gives a TBW of 16.6Hz.
Well, something fun to try. Build the sharpest 20kHz passband filter you can with whatever FIR design program you have available.Not one that goes 20kHz passband, 22kHz stop band, but as sharp as your software will design.Then filter castinetts with it and see what happens. Use ABX, of course. Just give it a try....What I'm thinking of and trying to build is a remez-designed filter with a cutoff frequency of 20kHz and a -90dB point of about 20050 (not 22050!) Hz.
Nothing bad happens AFAICT—I can't ABX it with castanets.
Quote from: bandpass on 10 April, 2015, 11:07:23 AMNothing bad happens AFAICT—I can't ABX it with castanets.Nice pics for illustrating the frequencies of ringing, better as this impulse crap we see elsewhere.Since the strong ringing happens at 20kHz it should be not touched by the filter of the DAC playing it back.If this isn't abxble what is?
Put another way it is impossible to know with any certainty the amplitude of a pure tone with only two regularly spaced points per cycle.
I don't want to hijack this thread to much but this talk about ringing everywhere makes me wonder.You can't argue much about it. To many audiophiles and people insist on this is what matters most with HiBitrate music.Archimago lately did some about how software is handling dsd conversion.dsd encodersUnfortunately on his blog everyone can add his own view. You see jk_audio rates the dsd conversation to 24/96 on the amount of ringing as it was the most well understood thing in audio.
If you take just 2 samples starting at 90° we get (+1, -1)
We could pull a ten year old off the street.
Quote from: xnor on 11 April, 2015, 01:39:26 PMIf you take just 2 samples starting at 90° we get (+1, -1)What if I had sampled that sine wave at 30 and 210 and gave you the data. How do you know how loud it was initially?
What if I had sampled that sine wave at 30 and 210 and gave you the data. How do you know how loud it was initially?
If it's one cycle of a sine wave, like you all keep talking about, that has an enormous bandwidth, not just a line at the frequency of the sine wave that was windowed.
Yes, of course, but only if the amplitude was zero outside that range of the single cycle.In the image I posted the sine continues with a dashed line in both directions, so interpret that 'single cycle' as just looking at a section of a pure tone of infinite duration.