In real music signals played back on real linear, time invariant audio systems, there is no "pre or post" anything. You have passband, stopband, and transitions between them which can be anything from very steep to very shallow. The only "ringing" that can occur in such systems happens when they are not properly damped such as for transition bands using high order filters. The whole notion of "pre" and "post" ringing has nothing to do with PCM A/D and D/A conversion and everything to do with impulse responses. And if you know anything about digital sample and hold or zero/hold circuits, you know that the spectral content is tightly controlled and band limited. There is no high frequency ringing if sampling rates are sufficiently high and the sampled content is sufficiently band limited with low order filtering circuits. If you impose band limiting as Craven allegedly did - rolling off frequencies at the upper end of the sampled content, it is possible to wind up with an impulse response of mush that attenuates both pre and post ringing spectral content while adding substantial phase delay (energy storage) with attendant post impulse oscillations. It's important to remember that we're dealing with continuous time invariant audio signals. There's no Hanning window. No time zero. Do yourself a favor and buy a cheap used 100mhz scope from Ebay or borrow one from a friend. Splice an RCA cable on your stereo and connect your scope to it while playing back music. See if you can find some impulse spikes that are in the microsecond range or lower in duration. Every now and then, it can be helpful to take a step back or two to look at the big picture. Context is everything and voltage/current spikes don't exist in a vacuum. There's a great big world of energy storage elements from tiny diaphragms in microphones to 200 gram 14 inch woofers and everything in between. None of this stuff responds in any meaningful way to stimuli that span mere nanoseconds or microseconds. And neither do our ears. Craven and the Meridian gang have had significant challenges to their professional reputations over the years. I'll leave it at that.
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+1 Erik “The full scope of it’s ambition, to correct all timing errors in the A/D, D/A chain and to do so using equipment specific data is monumental.” This is impossible. Everything is done in the studio to try and get it right with as little processing as possible. Basically every instrument and vocals are often recorded in a seperate chain to a separate track. No amount of clever processing in playback can magically fix a recording. You actually need master tapes with all the tracks separately to do anything really meaningful to improve a recording - anything else is just simple noise filtering and eq that affects the entire recording. MQA is hand waving nonsense. Stuart is so full of it. |
erik_squiresI'm afraid I can't follow your "everything and the kitchen sink" approach to constructing an argument Do not be afraid I have discovered that most often when an argument can not be followed/understood it is because the person putting forth the argument can not follow they're own reasoning, ideas and thoughts because if they could then everyone else could too. |
I subscribe to the philosophy at Schiit. There is no interest in MQA. Period. There is no interest in Class D amplifiers. Period. Same for DSD. Period. Jason and Mike make my predominantly Redbook collection of Music sound Magical. I would like to say that maybe a more expensive DAC might improve 24/96 Hi-Res files...but that isn't what I have collected. Make my CDs sound better, without charging me $25 per SACD or DSD download. Never heard MQA, probably never will. That's why I am listening to 16/44.1 CDs. Because I have already bought them. |
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