I’m getting whiplash from polarized — and shallow — opinions in the high end world.
In digital music specifically, I’m bothered by the hardened, and often fact-free opinions of both audiophiles and engineers, the latter who ought to know better.
As a design engineer and true audiophile and music lover, I’m a rare bird, sitting in both camps. I have learned (I don’t argue with reality) that things don’t always sound as they measure, and furthermore understand rational reasons for this (beginning with incomplete measurements). For this post I’ll try to avoid the quagmire of subjective thresholds and simply ask “where are the differences and what is possible?”
I’ll turn up the contrast. At one extreme are many who believe digital is fatally flawed, always has been, and cannot be cured. At the other end we have engineers who say “all is well, and if the bits are right, its perfect”. This is factually (technically) incorrect. I’ll touch on only one small aspect of why here.
I don’t want to boil the ocean. I only want to address the question of whether Redbook CD format is good enough for even highly discriminating music lovers and revealing systems, and if so, whether high res files and recordings can simultaneously sound better. I’ll touch other topics in future posts: (digital interface signals and their contribution to audio quality, why SPDIF and DSD among others are part analog) and other related topics.
My personal opinion is that, done perfectly (not in this world) RedBook — or 16bit, 44.1 k-sample, linear PCM encoding, is theoretically equal to and likely superior to any analog we are likely to experience – $5-10k turntables and all. The problem is, Redbook digital is rarely (ok, never) done perfectly. The “flaw” in the Redbook standard, again in my opinion, is that the sampling frequency chosen — for practical/cost reasons, makes it very hard for both studios and consumer playback equipment to perform ideal A–> and D–> A. These are analog processes folks.
The biggest problem in the standard itself is the 44,000-sample (don’t confuse this with 44 kHz analog signals) rate. This sampling rate was chosen to be more than 2X the highest audible frequency of about 20,000 Hz. Per Shannon’s math and Nyquist’s specific theorem, one must sample at **more than** 2X a frequency in order to faithfully reproduce a smooth, distortion-free “X Hz” tone – and all frequencies below it. Really, it can be perfect – but there’s a catch. If you have ANY — **ANY** — signals above 20 kHz that get into the recording path they can alias and play havoc with the recording, interfering down into the audio band. Plus, these sorts of non-musically related distortions are particularly annoying, leading in part to “digital glare”.
That’s one of the measurement flaws. All distortions are not created equal, sonically. Yep there’s good distortion and bad distortion, or at least bad and worse. This is understood well in music theory. A Boesendorfer or Steinway concert grand Piano is valued for its consonant harmonic distortions. So are (some) tubes. So distortion can be pleasant, or at least not unpleasant. Digital aliasing is not in that group – its just nasty. As is “slewing” distortion – and any odd-order, high-order harmonics. Back to the sampling frequency – to rid ourselves of aliasing nastiness, we must filter out 100% of that ultra-sonic stuff — the stuff above our cut-off frequency of 20kHz.
Ok, but i said it could be done. It can. In theory. The problem is, to get rid of everything above 20,000 Hz the standard only leaves us 4,000 Hz for filters. And good sounding, phase coherent filters typically work by roughly halving the sound every OCTAVE, not every 1/10 of an octave, which is what the standard leaves us, almost exactly. Bottom line #1: the filters used can be nasty. Bottom line #2: they are not 100% perfect so we typically get at least some aliasing. Maybe not much, but some. Note this is only ONE problem in the standard. But rejoice, there are real workable, solutions and they don’t begin with throwing away CD (16/44k redbook).
And this, in my worthless opinion, is why high res files (24/96 etc) sound better. They had WAY more headroom to work with for the filtering in the studio, and our home CD players have more space too. Furthermore, with 24 bits, engineers can miss their levels by a few dB and it all works out. And they can edit and make digital copies and still have perfection in the 16 or 18 most significant bits – which is still way better, on paper, than your mondo turntable – or mine (one of my collection is a Logic DM101 with a Syrinx arm and a Grado Reference, The other a Linn triple-play, if you care).
So we should quit worrying about format errors, and do two things:
1. Encourage studios to do the best job possible. Think they do? Listen to most rock, then listen to an old Verve two-track recording. ’nuff said.
2. Buy home equipment that gets the signal processing right. That’s another blog, but by this i mean low noise, low jitter, fast filter and buffer amps, and great power suppliers. Just like I built. Trust me, it works.
I hope you found this useful. When i have time to organize a complex subject I’ll tackle why the digital player and interface can make a difference. After all buts are bits. Its true… but that signal isn’t (just) bits. Intrigued?
CEO Sonogy Research LLC