Rev 8/29/26 (see notes)
Killer 16/50.4 non-oversampling digital decks with linear-phase anti-aliasing filters, such as the Mitsubishi X-80, were introduced in 1980, and were used for making digital copies of analog masters which were indistinguishable from the analog masters. Linear phase response is necessary to preserve the time-alignment of all of the frequencies in the input signal. The high sampling rate relative to the eventual 44.1 kHz standard made it easier to design and implement linear-phase anti-aliasing filters. Before Apogee introduced linear-phase anti-aliasing filters in September of 1985, 44.1 kHz recordings had a "smeared," "blurred," or "brittle" high end, a dry SQ overall, and mediocre imaging.
So, the albums copied with these highly transparent 16/50.4 decks were essentially perfectly preserved forever. These digital recordings have been used as source-material for LPs since then, although we've been given the impression that LPs are "all analog." In fact, digital signals ARE analog, consisting of two analog levels with brief analog transitions between them. So, claiming that LPs made from digital masters are "all analog" isn't a lie, although it is misleading.
In about 2005, asynchronous sample-rate-converter chips (ASRCs) were introduced, and it became possible to make 44.1 kHz recordings from 50.4 kHz recordings without converting to analog, and then to digital at the new sample rate. ASRCs mathematically simulate the D-A/A-D process in the digital realm, and they're so transparent that they're included in some pro-grade mixing consoles.
MoFi digital master scandal
Mobile Fidelity customers were outraged a few years ago when a MoFi employee admitted that they had been using digital masters. But the funny thing is that the MoFi customers were convinced that the LPs were plain old analog until they were informed otherwise. They should have been impressed by how "analog" digital can sound, instead of becoming outraged over something that evidently doesn't make any difference to the sound quality, other than to prevent degradation with time. But due to the placebo effect (we hear what we want or expect to hear, at least until the reality breaks through the mental block), a lot of them probably thought that their MoFi LPs suddenly sounded worse by being associated with the d-word.
According to MoFi, they used 4xDSD, where DSD is an analog modulation of a digital carrier, and it can be demodulated with a simple filter, so it's actually a form of analog that can be stored digitally. (The carrier is digital, but it would be filtered out.) However, I doubt that MoFi made 4xDSD recordings, considering how gargantuan such recordings would be. They probably said it because they knew that the digital they used (probably 16/50.4) was indistinguishable from analog, and that telling their customers that they used 4xDSD might eliminate their mental blocks so that they would hear what's really on the LPs. Even Michael Fremer, a staunchly anti-digital extreme high-end reviewer, believed that a digitally-mastered LP was made without any digital, and included it on his list of recommended LPs.
The notorious audio perfectionists Steely Dan used 3M decks (16/50.4), which they owned and reserved for their own use, to record their original masters in 1981 and to play them back to make LPs. This way, they obtained perfect net linearity, since 3M decks used the same DACs (probably mil-grade for reliability and stability) for recording (as part of the ADCs) and playback. They reserved these recording for their LPs, and dubbed them to 24-bit copies in 2006 with the latest ADC technology. So, they ended up with the equivalent of 24-bit copies made in 2006 from their original masters as they were in 1981.
MoFi's secret ADC
I once stumbled onto a reference to an ADC made for MoFi by Theta Digital, using multiple (probably four) BB 16-bit ADCs in parallel per channel to sample at a high rate and thus allow the use of a cleaner input filter. The high-rate digital was then digitally filtered down to 16/44.1 with a 20 kHz bandwidth. It probably sounded very good, and might have been used for some of their LPs without decimating the high-rate digital down to 44.1 kHz.
Chesky
Chesky had its 24-bit oversampling process at the dawn of the digital era, but it was patented and they kept it to themselves for use on their recordings made with a minimalist microphone set-up. The ADC was designed by Bob Adams, who later designed ADC chips for Analog Devices. Chesky's ADC used 1-bit technology, which generates relatively high output noise in response to clock jitter, but it was a discrete design, which made it easier to obtain high performance. Cymbals are particularly difficult for 1-bit ADCs to get right, but Chesky's reportedly did an excellent job on them.
Sony PCM-3324, Apogee filters
Sony's PCM-3324, a 16-bit, 24 track deck which used counter-type converters and sampled at 44.1 and 48 kHz, was a popular digital deck. It was introduced in its original form in 1981, and with the ability to cut and splice tape (to avoid the need for additional gear for editing) in 1984. But its input filters for the 44.1 kHz sampling rate weren't so hot. In late 1985, Apogee introduced linear-phase filters for 44.1 kHz non-oversampling decks, which dramatically improved their SQ. Different filters would have been used for a 48 kHz sampling rate, and Sony's stock 48 kHz filters might have already been adequate. TV audio, which is sampled at 48 kHz, should convince analog die-hards that digital has the potential to sound perfectly "analog." Its higher sampling rate facilitates the use of linear-phase filters and the content isn't degraded to reduce losses due to piracy, as it is for CDs. TVs also often use TI/Burr-Brown Segment-DAC chips (explained later in this essay), which
are inexpensive and known for their "analog" SQ. I suspect that they were designed for TVs, because the audio gear industry in general isn't interested in providing digital gear with "analog" SQ and low cost. TI's revolutionary DACs were introduced in 2000, but I had to discover them over 20 years later, by investigating why my Zen DAC V2 sounds so good for such a cheap DAC. TI could have advertised their DACs with small ads with an audio-pro's endorsement and a URL, but the nearest thing to an ad that I've been able to find is the claim included in their data sheets that they provide "excellent dynamic performance [24-bit resolution and 25-bit linearity] and improved tolerance to clock jitter."
Audio polarity
TV audio's specifications include audio polarity, which refers to whether a waveform is right-side-up or upside-down, which determines whether the speaker-drivers are moving the right direction at any given instant. Only once have I found a TV channel with inverted audio polarity, and I sent them a message and they fixed it. The Audio Engineering Society also has a polarity standard. So, clearly the audio police consider audio polarity to be a significant factor. I've found that the polarity is important if the record/playback chain is sufficiently coherent, but that even then it can be difficult to ascertain due to mental blocks caused by preconceptions about the SQ. These factors would explain why many people are under the impression that polarity doesn't matter. For years, I flip-flopped on the polarity of the Bitches Brew remix (available on CDs only), because my system had problems, and then when I finally eliminated the last of its problems (a defective USB cable driving my Zen DAC V2), I still had a mental block (something like a subliminal belief that Sony wouldn't mess with BB's polarity) which prevented me from hearing the truth. But then I lost interest in BB for several months, and then became interested in it again and realized that it has inverted polarity. [1] I suspect that the subconscious mind analyzes what we hear and draws attention to its conclusions by giving us urges to listen again, and then we hear it without the preconceptions. But if the system is incoherent/unfocused, neither polarity sounds right.
To invert the polarity of FLACs in batches, I use an Audacity macro consisting of the select-all, invert, and export-as-FLAC functions. Then I apply it to the files of interest by clicking on Tools, then Apply Macro, then Palette, etc. The results are placed in a folder named "macro output" in the Documents directory, although I assume that the destination can be changed. I then use a bulk file renamer to insert "-INV at the end of each filename (just before the extension).
High-res ADCs
The next major improvement in digital recording technology took place in 1993, with the introduction of high-end 20-bit oversampling ADCs. In 1997, 24-bit versions were introduced, and more advanced 24-bit ADCs were introduced in about 2005.
Turntables, etc.
After learning that Wirecutter recommended a Fluance TT, I looked into them and was impressed by their innovative, cost-effective designs, including such things as a heavy molded MDF chassis mounted on soft rubber isolators to kill vibrations, belt drive with regulated platter-speed for belt-drive smoothness and direct-drive's speed accuracy, and a $200 retail cartridge on a $500 TT. Well-Tempered Labs TTs supposedly have high-end SQ without the high-end price, but they aren't cheap.
Avoid linear-tracking tonearms, which are high-priced gimmicks with problems with no good solutions such as high lateral inertia and warp-wow. I designed a cheap one and wondered why there are no such designs, until I realized that the high price IS the idea. A simple pivoting tonearm with a fixed head-shell is the optimal configuration.
The great thing about CDs is that nothing about the transport matters other than being able to retrieve the data with sufficient accuracy to be able to correct read-errors, despite hucksters' attempts to convince us that CDs are just small LPs, and that their high-priced CD-transports improve their SQ by eliminating read-errors.
If you want to get the best SQ and longevity from your LPs, you'll need a deep-cleaning record-cleaning machine such as a Spin-Clean. Even new LPs should be deep-cleaned before playing them for the first time. For details, seek advice from multiple experts, because bad advice on this subject can be very expensive.
Some CD myths debunked
CD-grade digital has about 20-bit DR due to the use of narrow-band Nyquist dither, which has been in use since the 90s. Other forms of dither were used before that. The idea behind dither is to substitute noise for quantization error, which allows low-level detail to be heard through the noise. As with analog, low-level details can be heard through the noise, which in the case of narrow-band Nyquist dither is ultrasonic. Being able to hear low-level details constitutes higher DR. Another perspective is that the dither turns bit 16 into a pseudo-DSD channel, allowing it to represent smaller signals than it could as a PCM bit. It took a math PhD 75 pages to explain it thoroughly, so excuse me for not doing so.
The notion that "precision" CDs (such as SHMs and Blu-specs) sound better due to a lower error rate is a gimmick. They couldn't possibly have a lower CORRECTED error rate than standard CDs. (All digital storage systems include error-correction.) SHMs sound better because they're made from better masters, not because of a lower error rate. Blu-specs are typically awful, such as Bitches Brew, which was given an insulting "ear candy" EQ for the BS-CD (2008, 2013, 2023) to give the initial impression that precision CDs sound better. [1]
High-res myths debunked
Only recording engineers need high-res, so that they don't have to worry about setting the recording levels on live feeds precisely to avoid clipping or losing low-level detail, and can mix digitally without audible round-off errors. But not even they need the absurd sample-rates pushed by some outfits. Pro-grade-ADC designer Dan Lavry claims that anything above 60 kHz actually degrades SQ, although I surmise that 96 kHz isn't too excessive. Apogee concurs with Lavry. For more, Google "Goldmund Does high resolution audio sound better."
Modern ADCs use very high sampling rates at the input, partly to simplify input filters, but these samples are 5-bit samples of changes in the input signal between sample intervals, not 24-bit samples representing actual waveform values. The 5-bit samples are obviously converted to 24-bit samples in the ADC-chips. The basic idea behind using 5-bit samples vs. the 1-bit samples of early delta-sigma converters is apparently to reduce the amount of ultrasonic noise and jitter sensitivity, although I haven't been able to find any articles on multi-bit delta-sigma conversion. I have found ads for expensive books on the subject.
Digital interpolation filters
Digital interpolation filters (a.k.a. "oversampling filters") calculate and insert extra samples into the data produced by CDs, to turn the coarse "staircase" waveform from CDs into finer staircase waveforms, so that simple, cheap analog filters can be used to smooth them out completely without introducing any audible distortion. Without digital interpolation, it's expensive to create analog filters with a sufficiently steep roll-off and linear phase.
Back when the digital interpolation filters (DIFs) on DAC-chips sounded artificial because they were based on compromised designs to allow them to fit on the low-density chips of that era, 96 kHz recordings might have sounded better than 44.1 because they didn't need as much digital interpolation, so they weren't degraded as much as 44.1 by the lousy DIFs. High-end gear used natural-sounding "external digital filters," i.e. high-performance DIFs on separate chips. But external DIFs are rarely if ever used these days because the ones on audio-DAC-chips have gotten so good. Chord's DIFs supposedly sound better than typical DIFs on DAC-chips, but if so I would think that the likes of Audio Research and Bel Canto would use similar designs instead of settling for DIFs on DAC-chips. Audio Research got excited about a new tube socket design a few years ago, so they wouldn't put up with a bad-sounding DIF.
Why some CDs sound worse than the LPs
From the music industry's perspective, the problem with CD-grade or "better" lossless digital is that it's easy to copy it perfectly and anonymously, and it lasts forever. LPs, on the other hand, can't be copied perfectly and don't last forever, and people are leery of buying used LPs. So, to minimize financial losses due to piracy and used-CD sales, they typically don't put the best masters on lossless digital as long as the LP is selling well, which in the case of some LPs can be decades. Steely Dan reserves their best masters for their LPs. Tull will probably never release even a good version of any of their albums on CD, and you'll need the LPs for the best versions of Hendrix. The initial CD-release is typically the worst, and each new release is a little better, although the strategy varies from company to company, and album to album. In some cases, the audio is degraded by simply inverting it. In other cases, the digital master is converted to analog, perhaps processed or aged, and converted back to digital with an inferior ADC. I have a Bach CD which was degraded by mixing a key harpsichord passage in at such a low level that it's practically inaudible. I had a Brecker Pilgrimage CD which was so compressed and rolled off at each end that it sounded like a toy version. But it was an early CD-release, and the latest are probably good. So, there are many ways to degrade digital masters for CDs.
Lossy recordings
Lossy digital by itself might sufficiently degrade the content so that it might not be necessary to degrade it otherwise to protect it from piracy. So, the best masters might be released in lossy digital, which might explain why some people like lossy releases. HD Radio, which is lossy, sounds quite good if you don't listen too closely. Unfortunately, FM is a wasteland now that Sirius XM, which pays more to content-creators, is available. I haven't heard Sirius, but it might use lossless copies of the best masters. As I recall, Sirius XM receivers use TI/Burr-Brown Advanced Segment DAC-chips (explained later), which are used in some of the best high-end DACs.
Watermarked streams
Each stream is watermarked so that it can be traced back to the purchaser, so that if any copies show up for sale, they're easy to trace back to the stream's purchaser. So, piracy isn't a concern with streams, and good versions can be streamed, although the industry probably still reserves the best versions for LPs, to give us an incentive to buy LPs, turntables, etc.
Un-degraded oldies
But the best versions of albums that aren't big sellers might be available on CD. I've found a bunch of my old favorites, including Jean-Luc Ponty and Spiro Gyra, that sound just as good to me as the LPs ever did. But to find a good version of some CDs, I've had to buy a few versions. In some cases, there is no good version.
One of my favorites is the 1978 Pat Metheny Group album, a classic which I describe as new-age-soul jazz fusion. It's the one CD I kept when I got rid of all of my other CDs, but it wasn't a good version. The cymbals on the LPs always sounded flaky, even with high-end TTs. I later got the 2016 SHM-CD, but didn't appreciate it until I got the Zen and tried it a few times with inverted polarity (which didn't seem to make a difference at first, probably due to a mental block caused by the placebo effect). Then I listened to it with my headphones, and realized the cymbals are cleaner than they are on the LP, and this was before I replaced the defective USB cable driving the Zen. The cymbals are a type with unusually high frequencies, which are hard to put onto and get off of LPs. I suspect that the CD-master was made directly from the early-80s 50.4 kHz master, using an ASRC. It was inverted for the SHM-CD, but probably not for the SHM-SACD, which is expensive.
Steely Dan's Alive in America is available on CD only, and Dan pulled out all the stops for that CD. It was recorded and mixed in analog since digital mixing had problems back then. The mixer's output was digitized directly with an Apogee AD-1000 20-bit ADC, and Apogee UV-22, the original narrow-band Nyquist dither, was used to convert it to 16 bits with 20-bit DR. So, Alive is limited only by the playback system.
As mentioned previously, Dan's best digital masters, made with 3M decks in 1981 and transferred to 24-bits in 2006, are reserved for LPs. The 1999 CDs are derived from copies made in 1998 with an Apogee AD-8000 24-bit ADC from the 15 ips safety masters (copies of the original masters when they were new), which were preserved with TLC by Dan's main engineer Roger Nichols.
Gaucho was made without digital, and I assume that there was an analog safety master. Although the safety masters weren't necessary as backups once the 3M masters were made and duplicated, they ended up being used in 1998 for the digital masters for the 1999 CDs, when they had deteriorated sufficiently to release on CDs, and just after high-end 24-bit ADCs were introduced. The 1999 CDs are good, but the LPs are better. The 1999 Gaucho CD is very clean, but the last track (Third World Man) was apparently made from the 3M master, because it's a quantum leap better than the rest of the CD.
The best Katy Lied CD is the 1985, made without authorization by some renegade mastering engineers with a PCM-1610 with Apogee filters as soon as the Apogees were introduced. The 1985's were made from deteriorated 3rd-gen analog masters and sound lousy, but KL's master was Dolby encoded, so it still sounded good. (The multi-track and the first stereo master were dBx-encoded, but the first stereo master didn't sound right so some of Dan's engineers made another, but with Dolby.) When Nichols heard what had been done, he tried to get all of the 1985's out of circulation, probably because the combination of the Dolby-encoded analog master and the Apogee filters produced a digital master that was too good for release on CD. The first Dan CD-releases were made from the same analog masters, but a few years earlier with a PCM-1610 with the phase-distorting stock filters. These digital masters were also used for the Citizen and Very Best Of sets. The digital recording for the 1999 Katy Lied CD was degraded because it was too good to release as a CD.
I just recently heard Dan's Greatest Hits CD with my Zen and the new USB cable, and it sounds great. I don't know its pedigree, but there were plenty of ways to have made such a good digital master by then (1993), and there's no point in speculating about it. It's not as good as the LP I had in the 80s, which was squeaky clean, but good enough when played with good gear. My LP might have been made from a 3M copy of the original analog master, which would explain how clean it was. New LP-copies can be had, but they're hard to find and expensive. For the price of a few such LPs, you could get a premium Zen DAC and the CD, and would probably be satisfied with the SQ if you just listen to the music instead of analyzing the SQ.
Digital playback gear
Then the challenge is finding good-sounding digital playback gear that you can afford. I have a Zen DAC V2, and it's probably the last DAC I'll ever need. After hearing it, I couldn't stand my previous DAC, a Topping D10s.
The keys to the Zen's SQ is that it uses a "femto-clock" (a clock with sub-picosecond jitter specified in femtoseconds) and TI/Burr-Brown Advanced Segment DAC-chips (ASDs), each of which is a combination of an "R2R" DAC for bits 1-6 and a delta-sigma DAC for bits 7-24. The R2R DAC is split up into 63 nominally equal segments (which combined with zero gives 2exp6 = 64 combinations), each with a value of 1/64th of the total amplitude range. The 6-bit input is converted to a 63-bit "thermometer" code, in which each increase of 1 LSB in the input translates to another 1 in the thermometer code (like the mercury climbs in a mercury-based thermometer as the temperature increases), and each decrease translates to another zero. In a binary code, increasing the code can result in bits turning off, but in a thermometer code, they turn on only, which limits the number of codes to 64.
Splitting bits 1-6 into 63 segments allows the use of a technique which randomizes the connections between the thermometer-code generator outputs and the segment-switch inputs, which gives the segment-DAC 25-bit linearity by converting linearity errors into ultrasonic noise. It would be impractical to use this technique for more than six bits, so a DS-DAC, which is inherently linear, is used for bits 7-24.
The advantages of R2R/segment DACs is that they have low jitter-sensitivity, meaning that they don't produce much output noise for a given amount of clock jitter, and thus have a clean high end. DS-DACs have relatively high jitter sensitivity, but because the DS-DAC section in ASDs covers only the bottom 1/64th of the total amplitude range, its output noise is scaled down by a factor of 64 and is minuscule at the ASD's output. So, ASDs have the advantages of R2R and DS-DACs, without the disadvantages. One of the papers written by the ASD development team includes an oscillograph of an impressively clean -120 dB 1 kHz sine-wave generated by an ASD, back when clocks had much more jitter than they do now.
ASDs are good enough for Audio Research and Bel Canto, who make some of the best DACs in the business, without the stratospheric prices of some high-end DACs, such as dCS and Nagra's vast overkill status symbols. RME and Benchmark DACs, which are popular high-end DACs, use conventional delta-sigma DAC-chips and ultra-low-jitter clocks. RME makes no-nonsense gear for audio professionals, and their DACs have polarity controls that can be programmed into the remote. Benchmark's gear is a cross between high-end and pro-grade, and their higher-end DACs have polarity controls. I recall reading that Bel Canto DACs have polarity controls. Audio Research puts their polarity controls in their preamps. Some of Chord's gear also has polarity controls, but they're buried a few levels deep in menus, which is odd for a company that touts its digital interpolation filters, because polarity has more dramatic effects in higher-end systems.
For best SQ, I leave the Zen powered-up 24/365 with a cheap wall-wart with a 5.5mm OD/2.1mm ID barrel plug. Reviewers leave digital gear powered-up for at least 24 hours before reviewing it. It doesn't make a difference for some gear, such as my Topping D10s, but it does make a difference for the Zen. Not even the geniuses who design DAC chips seem to know why some digital gear sounds better if left powered up, but some of it definitely does.
The 20-foot USB3 cable between my PC and my Zen, which I bought from Amazon just for that purpose, would cut out from time to time. I assumed that the connectors were the problem, so I reseated them, and this usually fixed the problem, or so I thought. After about two years of this, the cable cut out, and reseating the connectors no longer fixed it. So, I replaced the cable with three short cables and was blown away by the improvement in SQ, compared to the SQ with the long cable when it was "working." As a result, I realized that the actual problem was that an internal connection was barely making contact, and that reseating the connectors was just wiggling the cable and restoring this connection to its normal data-degrading state. So, I recommend getting a USB cable tester to identify any cable problems quickly. Any standard USB cable that passes the test should sound as good as any other. It's OK to use long USB cables for CD-grade audio data because its data-rate is nowhere near the maximum data-rate for USB cables.
Music players
The only music players that have actual bit-perfect capability and can be trusted to retain that capability, without some agent of the audio-gear industry sneaking in any dirty tricks to interrupt playback or degrade the audio, are the Android players USB Audio Player Pro, Neutron, and HiBy. (When I went from MX-Linux 23 to 25, Deadbeef's performance went south, and it starts cutting out soon after starting to use it, although I've read that Strawberry has improved. I'll give it a listen some day.) I tried HiBy and gave up on it, but some people apparently love it. Neutron reportedly has optional transparent DSP including polarity inversion (actually a simple logical operation - just invert the code and add 1), and the Poweramp player also supposedly has transparent DSP but not a bit-perfect mode. A few player-developers have tried to use Android's built-in bit-perfect mode, but found problems with it, which explains the lack of players that use it. You'd think that Google, with all of the resources at its disposal, could devise a simple bypass for Android's audio system.
System maintenance
I have an old Denon analog receiver and live in a dusty area, so dust gets into the receiver through the ventilation slots on top of the receiver. This dust forms a layer on the circuit board, which acts as a parasitic circuit. Once a year, just after the monsoon season hits (which settles the dust and increases the humidity, and thus the parasitic effect of the dust-layer on the circuit board), I use an air-gun to blast the dust off the circuit board, and I reseat all of the connectors in the system, including the ones inside the receiver and the AC power connections. If I'm feeling energetic, I also cycle all of the breakers out to the power meter (which I do twice a year in any case). You could also use alcohol and a small paint brush to loosen any residual crud on the circuit board, and then let it dry and hit it again with an air gun. I never cease to be amazed at how this annual maintenance transforms my cheap system's SQ.
Speakers
My dream speakers are LX-Minis with digital crossovers with external Zen DACs. They're a sort of B&W Nautilus for those who can't afford to shell out about $80K for a pair of speakers, although the low end on LX-Minis is 40 Hz, which is the limit of most recordings. If I wanted to play LPs, I'd get the analog crossovers.
Ground loops
Everything in an audio system should be plugged into the same outlet, or the same outlet-strip etc. Otherwise, strange problems might occur. For example, I plugged my PC into one outlet and my audio gear into another, and the music player installed on the PC would cut out when the refrigerator turned off. Eventually it occurred to me to examine the set-up for ground loops, and found one. So I plugged the PC into the same outlet as the audio system, and haven't had that problem since then. Another strange problem also disappeared at the same time.
Notes
8/29/26 - Added material of potential interest to the Audio Polarity and Music Players sections, and a paragraph about Steely Dan's Gaucho (which evidently contains a cut from a 3M/LP master). Also made numerous tweaks not worth mentioning specifically.
8/27/26b - Added information about the first Steely Dan CD-release, the 1999 Katy Lied CD, and a reference to the Greatest Hits LP.
8/27/26 - Went through the entire article, clarified numerous passages, shortened and/or relocated others, and added some details.
8/20/26 - Added references to ASRCs (search for ASRC).
8/19/26 - Added details to the description of ASDs.
8/17/26 - Mainly replaced Note 1 after realizing that I was wrong about the polarity of the remix-CDs, as explained in the note, which I'm now certain are inverted. But I also made numerous tweaks and eliminated a claim that the 1999 Bitches Brew remix-CD was degraded by redigitizing it with an inferior ADC. After replacing the defective USB cable driving my DAC and listening to the BB remix-CDs with fresh ears after taking a break from them for about eight months and letting my preconceptions fade away, I realized that they have inverted polarity and that I like the 1999 more than the 2005 or undated Euro C2K 65774. I suspect that every remix-remaster has inverted polarity and some other subtle problem with the SQ, so that Sony can retain control over the best master and continue to sell us "new, improved" versions.
Revisions - After initially posting something, I inevitably realize that I forgot to mention something, that something needs to be rewritten, or that I included superfluous material that just wastes the reader's time. So, I edit it, and then post it again, and again realize that I forgot to mention something, etc. This can continue for quite a while, but if I didn't post it, I would never find the mistakes.
[1] The remix was made from the previously unused alternate 8-channel session master, which was 30 years old when the remix was performed, and in amazingly good shape for its age. The high end was still so extended that I have to use my headphones to appreciate it. Mark Wilder, the recording engineer, claimed that he made only an analog master, but I suspect that he also made a digital master (which is technically a form of analog - two analog levels with brief analog transitions between them), using a high-end 24-bit ADC such as the Apogee AD-8000, which was introduced in 1997, the year before the remix was performed. The resulting killer digital master (probably C2K 65774), one analog generation removed from the live sessions, could be degraded in various ways to make various releases over the years. For example, it could be converted to analog, processed in analog, or aged, to introduce analog types of distortion, noise, etc., and then converted back to digital with an inferior ADC. So, considering that each release could have been degraded, we still might not know how good the original stereo master sounds.
When I bought my Zen DAC V2, I also bought a USB cable to drive it. Over two years later, I discovered that the USB was defective. I had been assuming that any degradation caused by digital cables would be glaringly obvious. It would cut out once in a while, and I would reseat the connectors and apparently correct the problem, but the actual problem was apparently an internal connection that was normally barely making contact, and would periodically open completely, and reseating the connectors would wiggle the wire and restore the connection to its normal, data-degrading state. After over two years of this, it cut out, and reseating the connectors didn't do anything, so I replaced the cable and was surprised by how much the SQ improved compared to the SQ with the defective cable when it was "working." After replacing the USB cable, I assumed that I was finally hearing this CD well enough to make the call on the polarity, and declared that it definitely, absolutely has the proper audio polarity. Then I dropped the matter for about eight months.
Recently, I got the urge to resume scribbling about audio, and I eventually got around to the subject of BB remix-CD SQ. I realized that I still didn't have a good handle on the high ends of my favorites (1999, 2005, and the undated Euro C2K 65774), and blamed my tweeters, which have a hard time on a few other MD CDs. So, I donned my headphones (basic AKGs) and listened to the 2005, and soon felt that it just sounded wrong, and sort of dead, which to me is a clue that the polarity is off.
So, I flipped the polarity, and suddenly I felt as if I were listening in on the recording session. Then I checked the 1999 and undated Euro C2K 65774, and found that they also have inverted polarity. I could live with any of these CDs, although I'm leaning toward the 1999 although, as I recall, the high end is a little dry and dirty, because I became fatigued after listening to the undated Euro C2K 65774 for about 45 minutes, and the 2005 has a tinge of artificiality although the high end seems more detailed and liquid at first. It's as if there always has to be some audible imperfection so that nobody but Sony has access to the best master and that correcting the polarity can't fix all of the problems. Those who are unaware of the importance of proper polarity might buy each new release in search of satisfaction, assuming that their system has the right polarity. Those whose systems have the wrong polarity would probably find the remix-CDs more satisfying despite their other SQ issues.
The music itself is more enjoyable because it makes more sense and flows better with the right polarity, because the nuances are more audible. You basically haven't heard it until you've heard it with the right polarity.
So, I gather that when I got my system squared away eight months earlier, and thought that I was hearing the truth, I had a mental block that was preventing me from realizing that it has inverted polarity. Perhaps I subconsciously didn't want to admit that Sony's recording engineers could have gotten it wrong or that Sony would deliberately suck the life out of this masterpiece for business reasons.
No comments:
Post a Comment