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  1. I get it, I think. Are you able to give an example of non linear distortion, how it's measured, where it manifests. For example I don't think it would manifest under THD because jitter isn't an amplitude measurement.
  2. I thought I was understanding things, until now. So to paraphrase, what you're saying is that we can't get more than 6 microseconds of jitter within 6 microseconds of samples, because the discrepancy needs to be continuous to be audible? For example, if the oversampling was 6000x @ 1ns typical jitter then it would be audible in terms of timing? I thought jitter and timing were similar concepts, and that the total jitter per second exceeding 6 microseconds per second would be audible, because they would add up when our brain puts it all together. Now I'm even more confused. Slightly off-topic - how much jitter is audible? I know that changing the clock (in the same DAC) from a picosecond crystal to a femtosecond crystal improves audible jitter performance. No need for apologies, but I sure feel dumb right now.
  3. It depends on the clock, and the number of samples. A lot of DAC's have a jitter per sample in the nanosecond range in lab conditions, that puts jitter in the audible range even without oversampling.
  4. Well said, btw: I'd assume it's always perfect on the digital side, the damage comes in at the point the sample is converted to an electrical signal, such as in a zero-order hold it's waiting for the next sample but inaccuracies in the clock means it either comes too early or is delayed. But as long as it's a femtosecond clock, you can 8x oversample 44.1khz without the damage becoming audible. Hi-res without oversampling is definitely preferred, but they're harder to acquire than redbook. 95% of my music collection (over 2000 albums) is 44.1khz and I have no idea if the original recording was undersampled from 362.8khz or resampled from 384khz.
  5. Does that include oversampling? As I understand it you are not destroying the original signal, just putting additional signals in between. While resampling is reconstruction from scratch which, as you said, would damage the waveform. Oversampling will also require a very good clock, because with each additional sample you'll have a higher chance of introducing additional timing errors. What Grant said, while I agree, I hope it's not aimed at me? I said from the very beginning that my intention for this topic is mostly academic.
  6. I thought the difference between 44.1 and higher res is mainly due to smearing, i.e. inaudible content above the analog filter becoming audible when they are essentially reflected back and can impact anywhere from bass to midrange frequencies. So that's why we oversample and have a much higher digital filter where there are no content above the cut-over, but then jitter becomes an important consideration.
  7. Oh I'm not saying we should abandon high sampling rates for playback, there's still other advantages - just that I now understand Leshowitz's experiment better, the context of his findings and why 44.1khz is all we need for timing.
  8. I read what you said, mulled over it, and read it again and it definitely makes sense why 44.1khz is all we need. Thank you for the very thorough education, I have no more questions your honour
  9. Very nice article, I'm still going through it and the other paper you quoted will come later. So at this point my original point has been answered, most loudspeakers and companies do not design them to render to 10ns in temporal resolution. <-- I have graduated from saying "click"! (I know the Yamaha article says 6us, but doesn't quote references - or I've missed it).
  10. So we come back to my original point, which was that assuming Leshowitz is correct and that we can indeed hear a 10 microsecond click (not in frequency, but in timing), then a loudspeaker capped at 25khz in frequency response won't be able to fully render the click? Regardless whether the digital side is fed 44.1khz or 96khz. Again, I'm not saying I'm siding with Leshowiz, only that assuming he is correct in theory, and that I'm understanding his paper correctly.
  11. I'm having trouble wrapping my head around this. Let's say we are recording a 22.05khz tone, which is the highest frequency which 44.1khz can capture. So there is 22 microseconds per sample. If a click occurs 5 microseconds after a sample, and finishes in 15 microseconds, does this get captured in the digital recording? If so, how is this possible? A click at a duration of 15 microseconds is above the 22.05khz tone.
  12. Thank you for the informative response. I do understand the traditional terminology for phase is the degree of delay between the frequencies but didn't know how best to describe "short click", I generalised it to phase. Is "click" the correct terminology? The part where you said it's a myth to require high sampling rates to get the click right on the mark, are you able to elaborate further or point me in the direction of reading material? Thank you.
  13. It seems I asked the question poorly. I'm mainly interested in the drop-off of phase information above 20khz, tweeter only, crossovers are irrelevant. I think you've answered enough, and I've processed what you've said and I understand it better now. Thank you.
  14. Thanks @Ittaku that's the first concrete answer I've ever got, and makes sense to me. Is it also possible that phase information is conveyed differently in loudspeakers since they operate in the analog domain, so an activation (what's the right word here) of the driver diaphram may not line up with a single sample in the digital side, but could possibly activate on a sample above 25khz just because at that precise time, the analog signal is passed to the driver that contain that specific phase wave (if that makes sense). I'm not talking about ultrasonic frequencies, I'm talking about the audio timing. I agree we cannot hear over 20khz, but we can detect phase information in the microseconds according to Barry Leshowitz. I'm not saying whether he's right or wrong, but iif we can, then what phase information are our speakers rendering or attenuating.
  15. Did you read my original post? I know the basics, as I indicated in my post the sampling rate is double of frequency. That wasn't my question.
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