If a signal is band-limited to 20 kHz and then sampled at 44.1 kHz, there will be no aliases. Had there been content at 24.1 kHz, that would alias into the baseband, sure, but the premise was that the signal had been limited to 20 kHz.Ok, let’s break this down. You’re saying that the first alias doesn’t happen at 24.1kHz if a 44.1kHz signal contains no content beyond 20kHz? Because that’s indisputable and really easy to explain/demonstrate.
Ok, let’s break this down. You’re saying that the first alias doesn’t happen at 24.1kHz if a 44.1kHz signal contains no content beyond 20kHz? Because that’s indisputable and really easy to explain/demonstrate.
If a signal is band-limited to 20 kHz and then sampled at 44.1 kHz, there will be no aliases. Had there been content at 24.1 kHz, that would alias into the baseband, sure, but the premise was that the signal had been limited to 20 kHz.
This is trivial and somehow you completely missed his point.This is all trivial. If the signal is sampled at 44.1kHz and then reproduced, everything depends on anti-alias filter in the ADC and interpolation filter in the DAC. Properly anti-aliased signal coming into the ADC has no frequencies above 22.05kHz. It may have frequencies <=22kHz. If this properly antia-aliased and sampled signal is then reproduced by the DAC, everything depends on DAC interpolation filters, in case of S-D oversampling DAC, or on analog output filter in case of NOS DAC. If the output filter has leaks (not steep enough) above 22.05kHz, then mirror images (not aliases) are created at Fs - Fsig frequency. The mirror image is at Fs - Fsig frequency. Fsig and the mirror image create the beat frequency. This is bullet proof and cannot be doubted.
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Fsig = 21.5kHz and its mirror image at 44.1 - 21.5 = 22.6kHz
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This is trivial and somehow you completely missed his point.
I know and have designed a proper filter 2 days ago. This is out of the question/topic.My point is that you never know and the DAC must properly handle the signals <=Fs/2 (22.05kHz here). It must not create mirror frequencies and there is no excuse, otherwise you violate conditions of proper signal reconstruction. Go and learn how to design a proper filter and add it to Topping DAC filter menu, that would be the solution.
A bit like 320kbs MP3 which doesn't go beyond 18kHz. When playing back an MP3 at max.
Not completely on topic but somewhat relevant, one should never judge the bitrate or sound quality of these codecs by eyes. Someone in HA made this claim and of course, violated the famous TOS8. I played a game with that member, the results are hilarious. A very good example of expectation bias.When one would only play 44.1kHz MP3 (limited to 18-19kHz or so)
Most (decent) encoders will low-pass filter the audio at 16-18 kHz since what little spectral content exists above is barely audible (if at all). Removing the highest frequencies entirely allows all the bits to be used for the lower part of the spectrum. The resulting improvement in that range almost always gives a perceptibly better result. There is, however, nothing in the spec that mandates this behaviour.Ah.. I was looking at spectrum plots from MP3 created from RBCD and then converted back to WAV.
The files I played with all topped out sharply around 18kHz-19kHz (LAME enc, 320kbs, CBR, high Quality) where CD versions topped out at 20kHz - 20.5kHz.
Of course that will also depend on the content and perhaps VBR differs as well. Not all MP3 are created equal...
If I interpret this correctly (I am not a native English speaker), did you mean ADCs are supposed to be perfectly bandlimited? The answer is no, and in some cases even more severe than DAC filters.ADC would not let trough any step
This is a special step test signal made to test filters for various purposes (here i don't know why or the history of it ) .
But this kind of signal is "forbidden"
A very good article indeed .Here's my attempt at explaining it: https://troll-audio.com/articles/filter-ringing/
Sorry for my bad English this is band limited and of course "allowed" the single sample pulse exemplified mansr nice article is not something that's on your CD's or files and it can not get trough your ADC, bandlimited squares are ok I understand . But these filters are often demonstrated with single sample pulses , that are not bandwidth limited.It is not forbidden. You create perfectly band-limited square and see the rising edge of the step.
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But these filters are often demonstrated with single sample pulses , that are not bandwidth limited.
I do understand there is a technical need to know the filter impulse response . I wont do more off topic then (mod can move this if he feels the need) .
Sorry, are you being facetious? I did a decent amount of signal processing theory in grad school, I know what I’m talking about.
My point is that you never know and the DAC must properly handle the signals <=Fs/2 (22.05kHz here). It must not create mirror frequencies and there is no excuse, otherwise you violate conditions of proper signal reconstruction. Go and learn how to design a proper filter and add it to Topping DAC filter menu, that would be the solution.