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In Reply to: RE: Bad idea unless you can hear like a bat. posted by jmlpartners on June 06, 2009 at 13:10:36
jml, you must be aware that there are plenty of people that disagree with your theoretical take, right? I don't feel qualified enough to take a strong view, but there are plenty of knowledgable people who believe 192K sampling gives greater fidelity than 96K. It's certainly *widely* believed that 96K is audibly superior to 44.
Why?
- The Nyquist theory deals with continuous waveforms only - not stops and starts as occurs in real music. This point is critical.
- It is known that reproducing frequencies above 20 Khz has effects in the audio band, via intermodulation and by preserving transient fidelity. You are aware that super tweeters exist, right? (I don't use any myself and can deal with 3 dB down at 17 Khz...)
I will probably also record at 96K - though I'd like to try both.
And furthermore, my AN DAC at 16/44 is thus far the best digital I've ever heard - though I have no experience with hi-res PCM yet!
My point is just really that it is not nearly as cut & dried as you made it sound IMO.
Just because there are plenty of 'knowledgeable people' that disagree with me, I must be wrong. That’s the story of my life.
First of all, it is critical to understand that I was talking about the RECORDING sample rate, not PLAYBACK. The criteria for each in regards to audible fidelity are very different. There is no doubt 44.1K/16 is insufficient for both recording and playback. It is my contention that 88.2K/24 is sufficient for RECORDING LPs. Other than ticks and pops, there is no supersonic content of any consequence on an LP. The typical phono cartridge has enough trouble tracking 18KHz let alone 20KHz and beyond.
Nyquist theory deals with ALL waveforms. It is the practical implementation of the theory that screws things up. At every step, sampling theory demands band limited signals but, in practice, perfect band limiting it very difficult to achieve. In theory, 44.1K is sufficient to record all audio up to 20KHz but the filters necessary for the prefect band-limiting required by the theory have audible side effects. In the lab, with steady-state sine waves, 44.1K measured perfectly but not so with music due to the phase shifts and other anomalies introduced by the brick-wall analog filters.
That realization led to resampling and digital filters. With a digital filter, each output sample point is computed from a weighted average of the nearby input samples. The weighting coefficients determine the characteristics of the filter. In theory, digital filters are perfect and, in the lab with steady-state sine wave, they measure perfectly, as well. But music is not steady state and the averaging effect of digital filters takes its toll, such as with transients and every time there is a change in frequency.
The current state of digital audio means choosing from a spectrum of audible poisons. Audiophiles with a low tolerance for digital filter poison choose NOS and no filters. Others with a low tolerance for NOS poison choose resampling and digital filters. The reason many prefer hi-res recordings is they get the benefits of high sample-rate playback without the poison of resampling and digital filters. The cost is a huge dataset. Pick your poison.
My ideal is to start with lowest practical recording/storage bandwidth, 88.2K/24 for LPs and 44.1K/16 for CDs, and resample to a higher rate for playback without using digital filters. I have been working on this for many years and I now think I have the technology to do it. The first step will be to resample CD files to 88.2K/24 on the PC and feed both LP and resampled CD files to a custom DAC that will resample an additional 1x to 32x using both digital and analog techniques. The second step will be to resample both LP and CD files to 176.4K/24 on the PC and feed the same DAC. I suspect the poisonous effects of digital filters are most noticeable at lower sample rates: That’s why I think I can get away with using traditional digital filter to go from 176.4K to 705.6K and beyond. We shall see.
Mr. JML,
I have no idea who you are but your knowledge of things digital does seem to eclipse my own. If you read what I wrote to you previously I was only pointing out that there are those with completely different conclusions than your own. (It sort of seemed you might not have been aware of that, but obviously you are.)
I have heard a lot of digital setups and I can assure you that my AN DAC fed by the Mac Mini sounds divine - by this I mean not only pleasurable, but realistic in term of timbre, nuance, and dynamics, with no hint of 'digititus' at all. It has been my understanding that jitter is thought to be responsible for that hard, unpleasant digital sound, and since this is lacking in my setup, I have to conclude that it is not high-jitter. (While I may be relying on experience over theory, you do seem to be doing the opposite: have you listened to an AN DAC fed by a USB transport?)
I am open to the idea of using a physical disc transport as I did indicate. In fact, can you answer this for me: Is there some other way I could get 24/96 into my DAC? The chip will truncate to 18 bits but will take a 96 Khz stream, but the Hag USB interface will only take 48 Khz. Until AN Kits has their new USB interface there is no way to do it with USB that I know of. But there is no disc transport that I know of that could, say, take 24/96 AIFF files (such as those produced by a Masterlink with its CD24 burns) and transfer via SPDIF.
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