There's a lot to tackle this afternoon! I agree if people want to persist with this maybe it should move to another thread. Basically you guys seem to think digital is essentially perfect and I don't. Objectively and subjectively. I may change my mind with more experience but I doubt it. I am not trying to force my opinion on you. Obviously you are died in the wool, digital is more or less perfect, people. I don't want to ruin your enjoyment, I just don't agree. If you think AAC is perfect, or that high bit depth or sample rate is irrelevant, again, I simply don't agree. They're all very good though.
No tone controls in the tape demo? It was tape, right? Um...EQ? Calibrated to...um...exactly which test tape? SO little understanding of tape, the variables...etc.
The point here is that the ADC-DAC version was fed from the self-same master tape copy, already equ'ed to nominally flat in the tape machine and then out to the ADC. I agree there is only potential for things to get worse, but worse they did seem to get.
Finite FIR filter is windowed version of the infinite sinc. I don't see how that breaks causality. The spectrum is the product of the window spectrum and sinc spectrum. But if think digital is so inferior you can stay with your analog tapes.
Yes a time windowed FIR will be causal. But it is an approximation as you'd concede. Chord make a great deal of the FIR tap length issue and rightly so IMO. My favourite Nyquist filter algorithms (very limited test set) are the PMD ones. I actually seem to prefer fast roll-off to some of the 'clever' stuff.
But my point though is 1) that the sync x function is not paralleled in any real world acoustic effect or even practical analogue filter. Here are or step responses:
ht
tp://www.analog.com/media/en/training-seminars/design-handbooks/Basic-Linear-Design/Chapter8.pdf -see p25 and 31, 32 and others later.
No pre-ring on any of them. Later on there is an analogue anti-alias filter design presented for CD btw. Note this particular discussion regarding time symmetric impulse response is an issue with filter implementation as normally (read just about always) done in digital audio, not digital audio per se. 2) IMO digital improves as you go beyond 16 bit (I guess you think HDCD was a waste of time. Microsoft didn't, but customers, on the whole, did). and 44.1/48 ks/sec. And even if you are a little way below the noise floor (an analogue problem..), which would be an already excellent -96dB for 16-bit.
I actually don't have any analogue tapes! I don't even have any vinyl any more! Just CD, SACD, DVDA and downloads! I have Wolfson 24-bit and BB PCM R2R ladder based dacs. I prefer the (older) BB in my set up on CD. I like the ESS9018 and the new high end BB multibit/delta-sigma chip. The latter can give really good results
It's not a problem as long as our signal doesn't contain frequencies above nyquiest frequency.
It's not a problem if you want to play infinitely long sine waves, and that's why digital measures good on traditional parameters. Probably you have played with a spectrum analyser and adjusted the averaging number or time window. It's the same principle. Your sine wave spectral peak is ill-defined if you don't average enough cycles. That's not just SNR, it's maths. Well the converse is also true. Time and frequency are conjugate variables. Unless your waveform is being infinitely repeated, frequency domain limiting (admittedly of any form) results in temporal imprecision or distortion. This is just a byproduct of Fourier transformation. End of story. Music is not a repeating waveform.
So another point. Yes, analogue systems, or digital systems representing analogue parameters, all introduce time domain imprecision/distortion by means of bandpass effects. But the analogue forms are simple functions with everyday parallels, just like our mechanical electrical equivalents we were discussing with headphones. They are causal in time and infinite continuous in time. Digital ones are not. They are artificial in both respects. This is done because it is the easiest way to make a near-phase and -frequency linear anti-alias filter. In phase and frequency, they are indeed excellent. But actually, these parameters, carried over from analogue, are not really a good measure of time domain performance and hence of music waveform accuracy.
One thing I feel would resolve this for both of us would be a nulling scenario on music to look for the error. Feed both signals, time aligned and gain matched, into a really good differential amplifier. But again, this is actually really hard to do in practice because of time alignment and again, the filtering!