- Aug 20, 2022
@Geert "However, it doesn't make any sense to translate this time value into a frequency (the 100 kHz you referred to)"This does not mean, that we hear or need 100 kHz. 44.1 kHz sampling rate is perfectly able to reproduce that (and even less than 10 microseconds): https://www.audiosciencereview.com/...s/time-resolution-of-redbook-16-44-pcm.22102/
I distinguished tonal sensitivity from from timing sensitivity, that we do not hear 100khz as a tone but as a harmonic of audible tones. The link you've provided shows differences between left and right channel, which is only half of the problem given that there is still the capability of a given channel to consider.
You say this yet I'll bet you wouldn't buy for example a filter-less NOS dac over a delta sigma dac - because it sounds different.Fascinating. You have lots and lots of reasons why this device should make an audible difference, and yet it doesn’t.
@voodooless "That's why you need magnitude AND phase, as I mentioned.", "Distortion is distortion.. We can measure it, and it covers all kinds of distortion, also in timing."
The comment of mine you are quoting only mentions how only a frequency-magnitude can be unhelpful for the same reason that providing both a frequency-magnitude and phase plot can still be unhelpful. Take for example an IIR Bessel lowpass filter, a frequency chart will show roll-off, a phase plot will show nonlinear, yet the damping factor/square wave will prove superior to say a Butterworth filter of the same order. You can say that we should be able to infer that it is a Bessel lowpass filter that we're looking at even if all we had were the frequency and phase plots on hand, but that would be tedious in comparison to just being shown the step response.
"So we need oversampling after all?"
I was specific about how higher sample rate music would be provided natively (recorded that way) and not as a matter of oversampling, and if I were to say we need delta-sigma dacs after all, then I would be saying that we need oversampling after all.
"We don't listen to an impulse response. We listen to actual music"
All you're saying is that ringing is not detectible, which I would disagree with. Also, this sounds like when people say "I don't listen to measurements, I listen to music", which usually doesn't come unironically from this site.
"an oversampling DAC can and will reproduce the original waveform better than your NOS DAC, regardless of the sample rate. There is no disadvantage."
A NOS dac provided with natively high sample rate material and having an IIR lowpass filter vs a delta-sigma dac with the same material played into it: the delta-sigma dac's oversampling is redundant and results in higher than necessary ringing, the NOS dac filter's may have non-linear phase, but it's going to have the waveform with less ringing.
"No, the intermodulation is not generated upstream, it's directly output by the NOS DAC."
Hence my use of the word 'supposedly', it is an indictment of what is commonly conceived as the cause because what is upstream to the dac is only tangential.
"If you mean intermodulation products that reflect back to the audible part: obviously they are there, because a NOS cannot properly reconstruct the original waveform."
What do you mean by "reflect back"?
"What? I have no idea what you mean?"
My understanding of intermodulation is that the frequencies that would be labeled such can't count as harmonics. I have heard this worded the same way I put it, "not a part of the original signal". Of course that can be said about any form of distortion, but here it is used for emphasis.
"No, it just shoves the shit up in frequency, making it less audible.", "Sample rate != time resolution. As long as you think it is, you're lost..."
To the first comment: What does a delta-sigma do better than oversample? To the second comment: Yet I am supposed to think that a delta-sigma (read over-sampling) dac has better time resolution than a filter-less NOS dac as a result of something to do with the sample rate.
@Geert "Also, ITD refers to the difference in arrival time of an angled sound source between the 2 ears as a result of different path lengths. It's not a value that indicates absolute time resolution sensitivity, like an impulse being early without a difference in path length."
"An impulse being early without a difference in path length", Path lengths equals time difference. The relevance of there being two ears is that it provides further isolation of the sounds received by either, so that they do not interfere in space. Yet, they are only so isolated from each other before they are going to the same brain. The result is one waveform even if two ears are receiving two different sounds that are isolated before being summed.
@tmtomh "It always strikes me how much incorrect info about digital audio can be traced back to the stair-step fallacy."
If the graphs show stair-steps, why should I think otherwise? If the graph for dynamic range of an amp shows 70db, why should I think otherwise?
Nonsense.It's quite a good restaurant I hear too... although timing is quite important there, as in one must book in advance and there is strict table duration. Numbers are finite and the tables are not modular. The food is always on schedule, the gourmet chef ensures the timing is resolute. One must sample each item on the menu to understand the intricate nuance of the flavours... of course some excess sampling may be required for certain dishes. It really is on a grand scale.