What is with the high frequency noise above 60 khz running at about a 17 hz rate?
View attachment 28418
Looks like a side-effect of a filter. The period is likely the hop size of one filter step.
What is with the high frequency noise above 60 khz running at about a 17 hz rate?
View attachment 28418
Again, what about this shows us some momentum transfer above 20 khz is going on?
View attachment 28417
What is with the high frequency noise above 60 khz running at about a 17 hz rate?
View attachment 28418
Yeah, I've looked at that too.There's definitely content above 20kHz. But audible it's not at -68dB @ 20kHz. It extends to higher frequencies, but at levels below -100dB. I can post a WAV file with the frequencies shifted down, if you want to hear what this sounds like.
View attachment 28426
My hearing threshold is down by about -65dB at 20kHz compared to the high @720Hz (blue curve below). So a total of more than -130dB down from 0dBFS. Somehow I don't think I'll hear this at normal listening levels, but maybe if I crank up the gain by 240dB, like those lab rats, I might actually hear it for a split second. Until my hearing is destroyed.
More like a failed attempt on monetary transfer from audiophiles to MQA Ltd.Again, what about this shows us some momentum transfer above 20 khz is going on?
The filters are an essential part of theory and intrinsically tied to Nyquist theorem. If the original analog signal has frequencies higher than twice your sampling rate, and you don't apply a lowpass filter to remove them before sampling, then you will encode aliased artifacts. Then the reconstructed wave won't match the encoded wave. Theory says reconstruction is perfect only when the original wave was lowpass filtered before sampling.Here is how I see our discussion:
original signal --> ADC --> digital link/media --> DAC --> output signal
I thought we were comparing the original analog signal that enters input ADC connectors vs output analog signal on the output DAC connectors. In that terms original signal is what has been recorded. While I'm aware that AD process involves input lowpass filter, as well as DA process involves it on the output, I don't really see them as part of our discussion about Nyquist theorem.
The filters are an essential part of theory and intrinsically tied to Nyquist theorem. If the original analog signal has frequencies higher than twice your sampling rate, and you don't apply a lowpass filter to remove them before sampling, then you will encode aliased artifacts. Then the reconstructed wave won't match the encoded wave. Theory says reconstruction is perfect only when the original wave was lowpass filtered before sampling.
The relevance is this: Sergei seems to be claiming that removing high frequencies causes audible changes to the wave, even when the frequency threshold is above human hearing. This would violate theory, so the only way that could happen is if the encoding or reconstruction is not done properly. Or, if human hearing somehow has more acuity in the time domain, than in the frequency domain. If that were the case, after decades of research on human hearing I'd expect somebody to have discovered it by now.
You guys are using the word frequency to denote different things, hence the confusion. Krunok seems to mean the fundamental frequency of an arbitrary periodic signal. In the context of sampling theory, this isn't particularly interesting, which is why the rest of us are talking about the highest frequency component in a Fourier decomposition. Perhaps referring to it as bandwidth would be less confusing. To all but Sergei, that is.Nyquist theorem doesn't mention filters at all - I merely corrected your wording in the context of the theorem, nothing more.
You guys are using the word frequency to denote different things, hence the confusion. Krunok seems to mean the fundamental frequency of an arbitrary periodic signal. In the context of sampling theory, this isn't particularly interesting, which is why the rest of us are talking about the highest frequency component in a Fourier decomposition. Perhaps referring to it as bandwidth would be less confusing. To all but Sergei, that is.
Instead of only showing sample values you should also show the resulting wave form.
And, if you really want to make a point, you should also overlay the exact same down sampled waveforms.
How do you know what the original waveform looked like before it was recorded.
How do you know this is relevant to what you hear. I mean one can record in DXD format and look at the final bit values but can draw no conclusions on what bit levels sound like as the sample points are not the same as the waveform it represents.
Most certainly not R2R as that 'converts' sample points to sample and hold which is very different from the original. Even more so than DSD which at least makes 'smoother' transitions between sample points instead of relying on steep analog post filtering.