Bit-perfect Monk at Carnegie - like I never heard it before

9/5/24

While listening to my bit-perfect Deadbeef installation, configured as described in a previous post, I played Thelonious Monk Quartet with John Coltrane at Carnegie Hall, which I'd heard through Clementine driving my Zen DAC V2, and hearing it through Deadbeef was like hearing it for the first time. If you get any CD-grade Monk recordings [1], this should be one of them.

Anyways, I listened to music for about four hours, from about 3-7, without any interruptions from software. When I had the "buffering" problem mentioned in the previous post, I rebooted the installation, and it still had the problem. So, I suspect that it was triggered by date and time window, perhaps stored in a file which would be scanned periodically. If it happens again, I'll try changing the timezone, or system time, to see if it helps. But ultimately fixing the problem would require identifying the malware and removing it.

Notes

[1] Consumers don't need high-res, according to High-Resolution Audio: Does it Sound Better?​ by Goldmund Acoustic Laboratory, which is famous for its extremely expensive gear and would sell a super-expensive high-res DAC if they thought high-res sounded better. Recording engineers need high-res to efficiently produce clean CD-grade recordings. Mark Waldrep PhD, a.k.a. Dr. Aix, an expert in recording, was once a proponent of high-res for consumers, but now claims that CD-grade is all they need.

Something's rotten with Deadbeef, too

 As my luck would have it, Deadbeef developed what sounded like a buffering problem soon after I posted the previous blog-entry (about 4pm). By piping its output directly to ALSA, I had bypassed PulseAudio and Pipewire, so neither of them could be the source of the problem, unless one of them contains malware which takes effect whether or not they're being used.

But while listening soon after I got up this morning, Deadbeef didn't have this "buffering" problem for at least an 1.25 hours as of this writing, which is odd since I was listening to the same album yesterday when it was acting up. So, perhaps whatever causes the problem keeps track of the system date/time, and acts up on certain dates/days and/or times. If so, resetting the system clock might fix it.

However, I left the room for a while this morning, and it had stopped playing when I returned, although I hit play and it restarted at the beginning of the track and hasn't had any problems since then. Just strange.

Bit-perfect playback on Linux without interruptions, for at least a few hours

In the course of searching for information about preventing music-players and the underlying audio services/processes from altering data on its way from some music-file on a PC to an external DAC, I realized that "bit perfect" was the key to unlocking the information I sought. Using this term, I soon found a page entitled Bit Perfect Audio from Linux. After my previous unsuccessful experiences with attempting to attain bit-perfect playback with Linux, I was highly skeptical about whether this was actually possible, because I assumed that there was some music-industry conspiracy which had left no stone unturned in forcing us to pay big bucks for bit-perfect playback without interruptions ("musicus interruptus"). So, I suspected that the page's recommendations were no longer valid, but I decided to try one of them, namely installing the music player named Deadbeef (a name based on a hexadecimal code of some significance to coders) on a live installation of MX-Linux 23.3, setting its preferences as instructed, and then creating a Snapshot-ISO of the installation, which I then used to create another live installation which contains Deadbeef and the aforementioned settings by default.

The instructions for setting-up Deadbeef were apparently based on an earlier version of Deadbeef, so I've revised them for the version I was using (1.9.5):

(a) Click on the Edit, then Preferences.

(b) In the Preferences window, select the "Sound" tab and set "Output plugin" to "ALSA output plugin" and "Output device" to the relevant device, which in my case was the Zen DAC driver described as "iFi (by AMR) HD USB Audio - Direct hardware device without any conversions" (my Zen DAC V2 was plugged into the PC, so its drivers were added to the list).                      

(c) Then select the "Plugins" tab, click "ALSA output plugin" in the list until its settings appear in the window, and un-check "Use ALSA resampling."

After creating and booting the aforementioned live installation with Deadbeef, I put Deadbeef to use, and found that it's utterly simple - just drag the files or folders to be played onto the player, and the rest is obvious. At first, I wasn't impressed by the sound quality, but listening is very subjective and the sound quality you hear at first largely depends on what you expect to hear (the placebo effect), until reality overcomes your expectations.

So, I looked for some opinions on Deadbeef's sound quality, and got sidetracked by a bogus speed-error issue until I realized that it's irrelevant when using a DAC with an asynchronous interface (which probably includes any new DAC on the market), in which case the player's output data goes into a buffer, and the DAC, which has its own clock, pulls the data from the buffer as it needs it, in the course of clocking it out at precisely the right speed.

After eliminating the "speed-error" mental block, I listened some more using Steely Dan's Alive in America, which is my reference for cymbals, and realized that the sound quality is as good as I've ever heard through my Zen DAC V2, and that's very good.

But the icing on the cake is that there were no interruptions while listening for hours. There was an interruption at first, but it was apparently caused by something I did, and hasn't happened again.

So, this might be the solution to attaining bit-perfect playback, without interruptions, on Linux. But I wouldn't bet on it, because I thought I was in the clear with Celluloid on a full installation of Ubuntu Mate on a µSD card, until it started cutting out on the first day of the Labor Day 3-day weekend. If the interruptions are triggered by the system's calendar, they could still happen.

In case the interruptions strike again, I've ordered an Android tablet which I would use for running the Neutron player, which has bit-perfect capability and actually has a polarity ("phase") inversion function, which I thought the music industry had forbidden on mass-market gear. I plan on using the tablet to drive the 15-foot (or so) USB3 cable which I've been using between my mini-PC and the Zen, which works because the frequencies involved are relatively low (all my files are CD-grade). For best sound quality, I leave the Zen powered up constantly using a 5V wall-wart plugged into the back, so the tablet wouldn't have to supply any power to it.

IFI's Global Master Timing (GMT)

 

After giving up previously, I finally found some details on ifi's GMT clock besides the vague blurb included in the list of features on ifi's Zen DAC V2 page. The relevant page  was apparently part of an internal ifi presentation on various technologies used in ifi products, by their R&D division, for a business-oriented division.

The GMT clock was apparently developed for AMR's high-end gear, and has become ifi's universal clock, the best grades of which apparently have jitter on the order of 0.3 pS, and a very clean spectrum from 9 kHz to 15 kHz (which is all that's shown in the relevant plot, above, so perhaps it's the critical range). For information on how such low jitter can be obtained, see Analog-to-Digital Converter Clock Optimization: A Test Engineering Perspective.

Whatever the Zen DAC V2's jitter-rating is, its high end is as clean as any analog gear I've heard, which includes moving-coil cartridges and 45 RPM direct-to-disc LPs. This is partly due to the fact that it uses the TI/Burr-Brown DSD1793 Advanced Segment DAC, which has low jitter-sensitivity, low noise in general, and 24-bit resolution (see my Zen DAC V2 review for details). Naturally, the quality of the high end depends on the source material, and the cleanest CD I've heard is United We Swing by the Wynton Marsalis Septet.  

Reading the aforementioned presentation-page was the first time I'd run across the term "femto clocks," although I've done quite a bit of searching on the subject of jitter. It's another indication of how poorly the audio media has kept us informed of significant advancements in digital audio playback. But then considering that it's clearly downplaying the significance of audio polarity, it's apparently deliberately keeping us in the dark so that we'll flail about trying to obtain musical satisfaction from digital audio, and waste a lot of money on playback gear and recordings in the process. There are probably other, even darker motives, but what matters is that playing good CDs by means of well-designed digital playback gear which uses TI/Burr-Brown Advanced Segment DAC-chips, with a decent system otherwise, and with the correct audio polarity (although some CDs aren't very sensitive to polarity), is the least expensive approach to being able to listen to a particular piece of music when you're in the mood for it, and to obtain musical satisfaction from it.

Without further ado, here's the text from the web-page I found about GMT:


Femto Clocks – Picky about Phase Noise

Background

All Femto clocks are good, very good in fact. They exhibit jitter levels lower than most clock crystals which leads to better sonics. An oft quoted benchmark for accuracy is Femtoseconds (Fs) / parts per million [Fs for jitter, ppm for frequency error].

Back in 2008, before the word “Femto Clock” became all the rage, AMR developed a special type of clock in DP-777 [a $5K digital processor] as part of the “Global Master Timing” (GMT) and “Jitter less” technologies (why special? See below, because not all Femto clocks are the same).

We called it the GMT Clock platform (which is comprised of specialised hardware+software) as it is not just buying a “clock in a can” and job done.

Having worked with all sorts of clocks, including discrete, Rubidium, Superclocks and not the least Femto clocks over the years, we know them quite well.

All Femto Clocks exhibit excellent low phase-noise (measured jitter within the clock). However, as their origins lay in being part of SONET, the popular SONET targeted “Femto-clock” is less desirable as its best phase-noise performance is concentrated in the > 12KHz region (read: at the very top and way above the audible band, so benefits audio less).

As an example, this link highlights the use of Femtoclock technology in the telecommunications sector where they are spec'd for.

This [referring to the relevant jitter-spectrum plot at top of page] is an Optical Comm system (aka SONET - which is a Subset). We added the blue line to highlight the -70dBr region so that when referenced to the AP2 chart in the next section of the micro iDSD, it is more of an “apple to apple” comparison.

The spike at 50KHz is the "signal" As you can see, for quite a few KHz around this region, phase noise is low, this is what matters in this application.

However, the area around the green arrow is the most crucial human audible range of 20Hz > 20kHz where phase noise performance is less impressive in the region of -100dB to -70dB.

Explanation

Therefore, the key for AMR was to design a new system, the “GMT” clock platform which not only exhibits the lowest phase-noise in the crucial audible band, but offers precision (< 0.004ppm tolerance) adjustability with literally millions of possible frequencies  (as per the DP-777 "GMT" Technical Paper).

The GMT Clock system designed into the micro iDSD measures <280 Fs, comparable to many Femto-Clocks (because it was designed to give low jitter).

From the [GMT jitter-spectrum plot above], you can see that jitter in 9kHz > 15 kHz is very good, the micro iDSD noise floor goes all the way down to -150dB which is virtually across the board with no spikes.

How this benefits the user

Consistent, across the board negligible jitter means timing is supreme, with just the right amount of attack/decay and of course, tonal accuracy. We are really pleased with the very low jitter performance of the micro iDSD in the most crucial audible range – in fact, we would not mind if customers pit it against significantly more expensive DACs.

We hope you found this interesting as it sheds some light on the particular attention we have paid to parts performance and custom design in the micro iDSD (actually, we took it from the DP-777!).

Addendum: What about Rubidium Clocks?
 

[Above] is a chart of several types of rubidium clocks. What they all exhibit is many sharp spikes in phase noise. Even though they measure well, some down to -150dB, when they spike, noise levels jump up to -70dB to -90dB.

This is far from ideal which is why we have not used such clocks, neither in iFi nor AMR products. It all boils down to paying close attention to the specific clock/s used and its performance in the audible range.
 

Zen DAC V2 & TI/Burr-Brown Segment-DAC-chips

Keywords: "Burr-Brown, Advanced Segment, sound quality"

After the Zen DAC V2 is powered up for 24 hours [1] by means of a 5V 2A AC-to-DC adapter plugged into the Zen's External Power input, it sounds unbelievably good, with the advantages of analog and digital, and without the disadvantages of either. The source material seems to be the limiting factor, and while playing the United We Swing CD by the Wynton Marsalis Septet, I heard high-end detail that would probably impress any audiophile.

The Zen also has incredible low-level detail, due to the unique design of the TI/Burr-Brown Advanced Segment DAC-chip (ASD) used in the Zen (details below). As I wrote this, I was listening to the Monk Quartet with John Coltrane at Carnegie CD [2], and for the first time heard Monk quietly humming as he played.

After hearing the Zen, I can't stand my previous DAC, a Topping D10s, because I realized that it has no low-level or high-end detail, and produces lifeless "toy" audio with a noisy/harsh, artificially bright high end. This might be due to the Sabre DAC's relatively high jitter sensitivity (DAC output noise vs. clock jitter), combined with a relatively high-jitter clock. Regular delta-sigma DACs apparently require clocks with very low jitter to sound good, and such clocks aren't cheap.

After hearing CDs through the Zen, I realized that they have the POTENTIAL to sound better than LPs.  However, some copyright-holders want to retain control over the best versions of their recordings, so they reserve them for LPs since CDs can be copied exactly and easily. But most CDs are good enough for me, and some are fantastic.   

High-res sounds no better than CDs, according to High-Resolution Audio: Does it Sound Better? https://goldmund.com/does-high-resolution-audio-sound-better/​ by Goldmund Acoustic Laboratory. The higher sampling rates might have helped some ancient DACs with lousy digital interpolation filters, by reducing the need for digital interpolation, but only about 16 of the 24 bits are used for consumer releases. High-res formats were originally intended just for recording and processing.

TI/Burr-Brown Advanced Segment DAC-chips

What is probably the biggest breakthrough in digital audio playback history occurred in January 2001 with the introduction of TI/Burr-Brown model PCM1738 Advanced Segment DAC-chip (ASD). The Zen DAC uses the DSD1793 ASD, which is one of the most recent ASD-models. The ASD architecture (described below) is a more precise implementation of the Current Segment DAC (CSD) architecture, which is used in such models as the PCM1690, an 8-channel DAC-chip which according to Archimago's Musings' test of the Onkyo TX-NR1009 as a DAC, is an impressive performer.

But TI has never given this breakthrough the publicity which it deserves (such as small ads on audio-review websites with an endorsement from an audio expert and a link to a website with details), so that even now few people are aware of the existence of ASDs or CSDs, and even fewer recognize their significance. Their sound quality is typically attributed to their direct conversion of DSD (as if this would help PCM), by using names such as True Native or Bit Perfect. I didn't know about them until I got a Zen in 2023, wanted to know what makes it sounds so good, and investigated.     

The attached block diagram and waveform-illustration show the basic idea behind the ASD architecture, which is more properly known as the "Advanced segment DAC interpolated by sigma-delta," for reasons explained below. ("Sigma delta" is interchangeable with "delta sigma.")

The current-segment DAC (CS-DAC) in the block diagram (not to be confused with the current-segment architecture mentioned previously) is the actual DAC which converts digital inputs into analog outputs. It consists of 66 nominally equal current sources which are differentially switched between two outputs (positive and negative). The ICOB-decoder converts Bits 1-6 into a 62-bit  "thermometer-code" [3], and the delta-sigma modulator provides 4 such bits, but at 64x the sampling rate of the other 62 bits. When all are bits are off, the output (taken across the differential outputs) is -33 (-2.5 mA), when 33 are on, output is 0, and when all 66 are on, the output is +33 (+2.5 mA). So, it has a total of 67 levels, but since the 4 inputs from the sigma-delta modulator are switched at 64 times the rate of the others, they can represent any value between -2 and +2, with 24 binary-bit resolution, when filtered. This allows the total output to represent any value between +/-2.5 mA with 24-bit resolution.

Not all ASD-models use precisely the same configuration shown in the block diagram. For example, the PCM510xA family uses just a 32-bit sigma-delta modulator to drive the CS-DAC, probably because gear-manufacturers want to win the spec-war. However, the waveform-illustration probably still applies since it represents the fundamental concept behind every variation of the ASD architecture.

For purposes of the waveform-illustration, the ICOB-decoder (see the block diagram) is presumed to have an output of only 7 levels (0 and +/-3), when in reality it has 63 (0 and +/-31), i.e. 6 binary bits of resolution or (2exp6 - 1) levels.
 
The DWA block is a complex scrambler to randomize the connections between the CS-DAC's data sources (the ICOB-decoder and the sigma-delta modulator) and the DAC's inputs, in order to average-out mismatches between the CS-DAC's segments to obtain 24-bit linearity.

The characteristics of the on-chip digital interpolation filters (DIFs) [4] are also important, but the DIFs in the latest ASD-models sound great, although I'm not certain how they were in the first model, the PCM1738, introduced in 2001. The 1738 was used in some high-end gear, but it might have been combined with external digital filters.


Technical papers on ASDs

The attached waveform-illustration of the basic concept behind the advanced segment DAC is from a 2001 paper entitled "A 126dB D-Range Current-Mode Advanced Segment DAC"  by Norio Terada and Shige Nakao of TI-Japan. A 2000 paper, entitled "A 117dB D-Range Current-mode Multi-bit Audio DAC for PCM and DSD Audio Playback"  (apparently by the entire ASD development team) also contains useful information.  


A potential explanation for TI's cryptic ASD/CSD sales pitch

Every ASD/CSD-model's data sheet includes the statement "[model #] uses ... TI’s advanced segment DAC architecture to achieve excellent dynamic performance and improved tolerance to clock jitter." I'm not entirely certain what they mean by "dynamic performance," but dynamic range specs are listed under that heading in ASD data sheets. It might also have something to do with the ASD's 24-bit linearity and its clean transitions from one sample to the next, with minimal switching- and jitter-related noise in the output, as a result of the CS-DAC's number of levels, its differential nature, and the design of its differential switches.

The 2001 paper includes a jitter-sensitivity analysis which indicates that the jitter sensitivity of a multi-level DAC is approximately proportional to the full-scale range and inversely proportional to the number of levels, which in the case of the ASD's current-segment DAC is 67 levels (0 and 33 levels above and below). The paper doesn't mention the sigma-delta modulator's jitter-sensitivity, but it has five levels, and its amplitude-range at the CS-DAC's output is only about 6% of the total range (4 levels of 66 total), so it probably doesn't make a significant contribution to clock-jitter-induced noise at the ASD's output. I suppose that its small amplitude-range also gives it tighter control over low-level signals compared to a full-range DS-DAC. This, combined with its low clock-jitter-induced noise, would explain the incredibly clean low-level detail I've heard from the Zen.

The 2001 paper also includes a photo of an impressively clean -120dB, 5 mVp-p 1kHz sine-wave, produced by an ASD and displayed on an oscilloscope. The clock-jitter isn't specified, although the paper mentions that 200 pS was typical and 100 pS was about the minimum in audio gear at the time. Commonly-used modern crystal oscillators apparently have about 10 pS. RME's Steady Clock is rated at 0.1 pS.


ASD-endorsements by high-end audio-gear designers

From Bel Canto's chief engineer John Stronczer in the Stereo Times review of the Bel Canto DAC 3.7 :  
 
"At Bel Canto Design, WE STICK WITH THE CLASSIC PCM1792 DAC BECAUSE IT REPRESENTS THE BEST DAC PERFORMANCE ACHIEVABLE TODAY. This holds true for every aspect of the DAC - noise, distortion, dynamic performance and even the way the analog output electronics interfaces with the PCM1792 DAC is superior to other options."

Thorsten Loesch, the head of R&D at AMR/ifi when the Zen DACs were being developed, endorsed ASDs here (see the Reader-view page): 

"R2R DACs also have problems with low signal levels [due to linearity errors, which severely distort low-level signals], which is what Delta Sigma DACs avoid. Conversely Delta Sigma (including DSD) technology performs relatively poorly at high signal levels in comparison to multibit solutions.

That’s why we prefer hybrids; multibit architecture for high signal levels, (the so-called upper bits or the MSBs), and high speed (DSD256 or higher) Delta Sigma topology for lower signal levels. The Burr-Brown DSD1793 we’ve been using so much is one such hybrid and one of the best options for us."

According to "MLGrado" on the regular (non-Reader View) version of the aforementioned web-page:

Essentially the BB DSD1793 behaves like an R2R DAC.
[...]
The Burr Brown 'segment DAC' ... was thought by its makers to be A BREAKTHROUGH ADVANCEMENT OVER PURE R2R DACS OR PURE DS DACS. IT AIMS TO GIVE YOU THE BEST OF BOTH WORLDS, WORKING AROUND INHERENT WEAKNESSES. [emphasis added]

So one may indeed consider going to R2R a step backward. In any case, you would not have iFi's TRUE DSD conversion. R2R must convert DSD to PCM.

[end of excerpts]

ASDs are used in everything from Sirius XM receivers, CD players, DVD players, TVs, etc., to extreme high-end DACs. My cheap Vizio TV's "analog" sound quality puzzled me for a long time, but now I realize that it probably uses ASDs. Audio Research's latest DAC, the approximately $8K DAC9, uses the PCM1792A ASD. Bel Canto also uses them in their latest gear. In a Steve Hoffman forum entitled "Enamored by the Burr-Brown PCM1796 DAC chip," someone claimed that a Pioneer DVD player's audio section with ASDs blew the doors off of an expensive discrete R2R DAC. A Zen DAC Signature V2 review on Amazon claims that the Signature version sounds better than a $1500 DAC which uses R2R chips. A lot of companies with a reputation for audiophile sound on a budget, such as Music Hall, Musical Fidelity, Denon, and Onkyo use ASDs in some of their gear. Yamaha's WXAD-10 streaming-DAC uses PCM5121 ASDs and reportedly sounds "superb" and "engaging." Pro-ject uses ASDs in their recently-introduced CD players. The $130 Soundavo HP-DAC1 DAC-amp also uses ASDs and got a rave review on Amazon by a 70-ish vinyl junkie who wasn't impressed by digital previously. So, it's not very difficult to find DACs, streamers, receivers, etc. which incorporate ASDs and sound great. But almost nowhere, except in TI data-sheets for individual ASD-models, are they identified as ASDs, which is their most significant aspect as far as sound quality is concerned.

Regular sigma-delta ADC/DAC chips are used in studio-grade ADCs and DACs, so the standard sigma-delta technology can obviously provide reference-grade sound quality. (There apparently are no ADCs which use ASDs, which might mean that the ASD architecture is incompatible with the requirements of ADCs.) However, studio-grade ADCs and DACs typically cost thousands of dollars, partly because they have clocks with extremely low jitter. The $1300 RME ADI-2, which is popular among recording engineers, has less than 0.1 pS of jitter and sounds great with either AKM or Sabre DAC-chips.  

Zen's Output Stage

The Zen's output stage, as far as I can tell from various vague descriptions, consists of an OV2637A amplifier module, and associated circuitry to configure the module as an I-V converter and anti-aliasing filter in the Zen DAC. The module includes four single-ended amplifiers, each consisting of a JFET-input op-amp followed by a buffer such as a TI BUF634A (which can drive headphones), and each of which is used as one side of a differential pair, forming two differential pairs. It takes the form of a tiny circuit board with pins protruding from one end, and chips attached directly to it. Some of these chips are surface-mount devices, and others are bonded in place and connected to circuit-board traces via wire-bonds. The wire bonds and associated chips have to be protected, such as by epoxy encapsulation or with a cap bonded to the circuit board.

To provide sufficient supply voltages for the op-amps and buffers, the Zen has a voltage-doubler (a flip-flop, diodes, and caps), and the resulting dirty 10V supply is regulated down to about 9V, which is capacitively bypassed to provide high instantaneous current capacity. With a 9V supply, the maximum single-ended output would be about 8.5Vp-p, and the maximum differential output would be about 17Vp-p, which is a hefty signal from USB power or a 5V wall-wart. The Zen has single-ended and differential outputs for headphones and amps, so it's basically pro-grade.

Zen shows off good CDs and exposes flaws in bad ones

Esperanza Spalding's Radio Music Society CD didn't particularly thrill me when I listened to it through my delta-sigma-based DACs. But when I played it through the Zen DAC, I was amazed by the CD's sound quality, and mesmerized by the intricate music and Spalding's enchanting voice.

Another great CD is Alive in America by Steely Dan, whose main recording engineer Roger Nichols used it as an opportunity to prove how good he could make a CD sound, since it's a live CD and he didn't have to reserve the best version for LPs. So, he recorded and mixed it in analog, digitized the mixer's output with an Apogee AD-1000 (20 bits, 44.1 kHz), and used an Apogee UV-22 to convert the 20-bit data to 16-bit data that sounds like 20-bit data. Sign In Stranger sounds particularly good.

However, the Zen's excellent low-level detail is a two-edged sword, because it makes any flaw stand out. For example, the high end of my Kamakiriad CD (Reprise 9 45230-2) vaguely bothered me through the D10s, but through the Zen is obviously deliberately degraded. The LP probably has a great high end, and perhaps a more recent CD-release has a better high end.

Miscellaneous audio information

There is no need for precision CDs or expensive transports, due to the error-correction system built into the CD record/playback system. Precision CDs might contain better recordings than the ones used for regular CDs, because the superior sound quality can be attributed to the physical precision. But if the copyright holder reserves the best versions for LPs, no digital release will sound as good as the LP, although MP3's might be made from the best version since they use lossy compression.

Separate CD-transports cannot possibly reduce the CORRECTED error rate compared to that obtained from a cheap transport in a CD player, which is a few ppB for CDs in decent shape. If you doubt that CD read-errors can actually be corrected, explain how software can be stored on CDs (simply re-reading the erroneous data won't work). Using separate transports just adds compromises or cost to the task of synchronizing the transport's clock and the DAC-chip's clock. Ideally, the clock should be next to the DAC-chip's convert-command pin, to minimize jitter in the DAC's convert-command, and the clock's signal would be sent to the transport, which can tolerate higher levels of jitter. Any buffer-memories between the transport and the DAC-chip could also tolerate a higher level of jitter, as long as the DAC's data-inputs can tolerate the jitter in the buffer's output.

CD treatments don't reduce the corrected error-rate - they reduce laser-servo hunting, and thus reduce laser-servo-induced noise on the transport's power supply. The only CD players to benefit from CD-treatments were cheap early players with poor isolation between the transport power supply and the analog power supply, which allowed some of the laser-servo noise to get into the output.

Software is stored on CDs & DVDs, and it can't have any errors, so it's obviously possible to truly correct CD read-errors. Software has an extra layer of correction compared to audio, but it's rarely needed.

Some CDs, although very few, have inverted audio polarity, probably deliberately, in an attempt to prevent us from being able to rip the best version. So, if a CD just doesn't sound "right," try flipping the polarity. Flipping polarity doesn't make much difference if a recording is incoherent or has inconsistent polarities within it, or if the playback gear is incoherent. Examples of CDs with inverted polarity are Night by John Abercrombie, the CD layer of Michael Brecker's Pilgrimage SACD, and Bartok String Quartets 1-6 by the Alban Berg Quartet. Audio polarity can be inverted by flipping the polarity of each speaker connection, assuming that the speakers are passive, or by ripping the CD and using an audio-editing program to invert each file.

Some high-end gear, such as Audio Research preamps and Benchmark DACs, have polarity/invert switches, as does all sorts of pro-grade gear. Some DAC-chips, including some ASDs, have invert-functions. The fact that so little consumer-grade digital playback gear has polarity-control capability, although it would be cheap and easy to add it, supports my suspicion that inverted polarity is being used as a means of degrading the sound quality of some CDs.

To get the best sound from your system, periodically disconnect and reconnect all connections, including internal connections, AC connections and breakers (turn off everything that uses a significant amount of power before cycling breakers). Also use canned air to blow dust out of ventilated components, because dust acts as a parasitic circuit, especially in high humidity. In severe cases, it might help to scrub with alcohol and something like a toothbrush around components to eliminate all parasitic paths.

The optimal turntable configuration is belt-drive with a pivoting arm with a fixed head-shell. Direct drives have a feedback loop and their speed oscillates around the ideal speed, and tangential tonearms are a scam. Use a record cleaner such as a Spin-Clean, even if you have to scrimp on the TT or exceed your budget, because a really clean record sounds much better, and you can't undo record wear.

Notes

[1] Experts recommend leaving digital playback devices powered up all the time (unless they use a lot of power or have tubes which can't be shut down independently of the rest of the circuitry), and letting them warm up for at least 24 hours before judging them. For example, see Audioquest's technical paper entitled Evaluation of Digital Devices.

Any functional 5V 2A wall-wart with a 5.5mm OD/2.1mm ID barrel connector output can be used as a power source for the Zen. (See Prodigit's page entitled "How to test Noise from the Output of Power Supply" ) Some claim that higher-quality supplies improve the Zen's sound quality further, which I suppose might be possible.

When power is applied to the Zen's External Power input, the LED next to the input glows, and the USB connection isn't used for power. The LEDs on front indicate whether the USB input is active.

[2] The recording on the Monk Quartet at Carnegie CD was made on 11/29/57, using tube gear. ("Tube sound" strikes me as very coherent and alive. According to the VTL site, tubes sound better than other devices because they have a more linear gain, and need less feedback.) The tape was poorly labeled, put in storage, and forgotten, until it was discovered 48 years later in 2005, and digitized and digitally processed (see Recently Discovered Jazz Jewel Restored by Hip-Hop Legend). The resulting sound quality is amazing considering the age of the tape when it was digitized, and the performance was excellent.

[3] In a thermometer-code, each single-level increase is achieved by simply activating another bit, so that as the signal-level increases, more bits turn on, and none off, and as the signal level decreases, more bits turn off, and none on. Likewise, in a liquid thermometer, the liquid rises and falls with temperature, with no gaps.

[4] DIFs basically calculate samples to be inserted between the samples provided by a CD, for example, thus reducing the size of the steps in the "staircase" waveform at the output of the DAC, so that a simple, clean, and cheap high-frequency analog filter which doesn't affect the audio range can completely filter/smooth out the staircase.

Were Steely Dan's actual analog masters digitized in the early 80's with a 3M deck?

Rev 2/28/22 (see Notes)

Steely Dan has a reputation for being audiophiles, and Roger Nichols, their main engineer, was a proponent of digitizing analog recordings to preserve them. So, I find it hard to believe that they didn't digitize their analog masters (either the multi-tracks, the original stereo masters, the  safety copies which Roger Nichols made when each album was new [1], or all of these), with a good ADC until 1998.

In fact, an article entitled Roger Nichols: Digital-To-Digital Transfers from the May 2006 issue of Sound on Sound indicates that Nichols used a 3M digital recorder to copy analog multi-tracks in 1982, and that he was transferring at least one of them to Pro Tools in 2006, which just happens to have been the 30th anniversary of Aja, when the legendary Cisco LP was released:

"I am currently transferring digital multitrack tapes from early (1982) projects into Pro Tools for surround mixing. I remembered some of the problems with early recordings that related to early A-D and D-A converter designs. During these transfers I wanted to correct any of the early shortfalls, if possible. Even though they were 16-bit recordings, transfers to 24-bit would help preserve the accuracy of the original recordings.

"These early 3M digital 32-track machines did not have digital outputs, so the transfers were to be made via analogue cables into new 24-bit converters [something  more advanced than the Apogee AD-8000 8-channel 24-bit ADC which he obtained in 1997/8, I presume]. There was no such thing as a 16-bit converter when the 3M machine was designed, so they used a unique combination of a 12-bit converter with an additional four bits of an 8-bit converter for gain ranging. [I gather that this means that 4 bits of the 8-bit DAC were used for controlling the reference voltage on a 12-bit multiplying DAC.] This required a very expensive HP spectrum analyser to set the tracking of all the converter elements.
[...]
Converter Tracking

"Let's compound the issue a little. A-D converters have the same problem. The error for each bit during the recording is added to the error for the bits during playback. In early digital machines they hand-matched A-D and D-A converters to match closely to get the best sound on each track. If you had to replace a converter, you were in big trouble unless you replaced both with a matched pair. SINCE A-D AND D-A CONVERTERS BASICALLY WORK THE SAME WAY, SOME MACHINES USED THE SAME CONVERTER FOR RECORDING AND PLAYBACK TO AVOID TRACKING PROBLEMS. [In other words, the linearity error of the recording was canceled out by playing it back through the same DACs which were part of the ADCs which were used in the A-to-D conversion-process. There would still be some quantization error of less than 1/2 LSB, but this would be way down in the noise, and dither might not have been required. The catch is that this required the same 3M deck which was used for making the recording to be used for playing it back, perhaps decades later, which would have been risky because converters can fail. So perhaps two or three recordings were made on separate decks running in parallel to ensure that at least one of the decks would survive for a few decades. It would have been an expensive approach, but it would have provided extra security because someone couldn't steal a tape and play it on just any deck without losing sound quality. It was if the ultimate sound quality was encrypted, and that the key to obtaining it was to play it on the same deck that was used for recording it.]

"There were linearity problems with the 3M machines, but you could set the D-A tracking to match the A-D tracking so that the throughput of each track was linear unto itself. This meant that what you recorded on a track was what you played back on that track. This would be good enough for the transfers I needed to make. I DID HAVE A DIGITAL INTERFACE BOARD THAT I BUILT FOR THE 3M [2], but if I'd used that, the digital transfers would have had the non-linearity of the A-D converter without the correction of the D-A converter. So, analogue transfers it was to be. [Well, actually, he wanted to make a 24-bit copy, and playing it back on the deck which was used for making the recording (i.e. each channel was played back through the same DAC which was used in that channel's ADC), and filtering it with the 3M deck's (analog) anti-aliasing filters [3] was the best way to precisely re-create the original waveform so that it could be digitized with a 24-bit ADC, and the resulting digital recording could be encrypted for security.]

So, as far as I'm concerned, the evidence is sufficient to conclude that Nichols made 3M digital copies of Dan's multi-track and original stereo analog masters back in the early 80's, and that they were converted to 24-bit copies which are being used for LPs. Any evidence or arguments against this conclusion can probably be explained away. For example, the fact that they made a surround-sound version of Gaucho, supposedly from the analog multi-track, and then never got around to the other albums, might have just been intended as a cover story for the existence of a 3M copies, and perhaps they concluded that it wasn't worth the effort to make surround-sound versions of the rest (not even their latest albums were released as surround-sound versions, as far as I know). If they want to retain control over those recordings, it's their right, and I'm satisfied with the 1999 Aja CD-release played on my Topping D10s (an awesome-sounding cheap DAC) and the knowledge that if I were to shell out $500 or so for a decent belt-drive turntable/preamp with a decent pivoting tonearm [4], about $100 for a good record-cleaner, and $50 for the LP, I could hear Aja in its original splendor, or even better, because the original recordings will last forever, and LP-recording and playback continue to improve.

Notes

Revisions

2/24/22 - Revamped after finding the 2006 Sound on Sound article by Roger Nichols using 3M decks to copy analog multi-track masters, which eliminated the need for a lot of the circumstantial evidence and speculation in the previous revision.

2/25/22 - Corrected a few conceptual errors, added some insights, and smoothed out some rough spots.

2/28/22 - Revised Note 4.


[1] Nichols stored the safety masters (15 ips analog copies of the original stereo analog masters) in his personal archive, and never played them, except for Aja, once, to make the 1982 digital master with a non-oversampling 16/44.1 Sony deck with the notorious dry/smeared-sounding input filters. (Some tracks from this version of Aja can be heard on the Very Best Of Steely Dan CD, a compilation made from the 1981/82 digital recordings. The Greatest Hits CD was apparently made with Apogee AD-1000 20-bit ADCs from a good analog master, and sounds good, although the analog master was created in 1978 and it was apparently digitized in 1993 when the AD-1000 was introduced, so it's not as clean as the LP-copy I had in the mid-80's.) Nichols could have used the 44.1 kHz oversampling deck which JVC introduced in 1982, and which made great recordings, but Dan probably didn't want to let such good digital copies out into the wild. The 1985 digital masters were apparently made with a Sony 1610 with Apogee input filters from the same analog tapes, which by then had seriously deteriorated so that the high end is little but a dry sheen, except for Katy Lied (dBx-encoded) and Gaucho. The Gaucho digital master has a slightly high speed, so that the slow cuts sound rushed, and the Aja CD has a peaky high-end boost which sounds dirty and artificial. The MoFi Aja and Gaucho CDs (which I haven't heard) were made from the same digital masters, and according to Nichols, in an article in Metal Leg 18, Gaucho has the same speed error as the commercial release. However, MoFi might have used different EQ on Aja. MoFi got their own oversampling ADC in 1988 from Theta Digital. It used BB ADCs in parallel to sample the input at presumably 176 kHz, and Motorola DSP chips to implement the digital filter that reduced the sampling rate to 44.1 kHz.

[2] Nichols obviously broke out the digital signals so that he could duplicate digital recordings without converting them to analog. 3M probably sold accessories for this purpose, but Nichols, who was famous for creating recording-related gear, might have found a better approach, or at least a less expensive one. If the duplicates were played back on the original recorder, their errors would be canceled out.

[3] It was easier to design anti-aliasing filters with a linear phase characteristic for converters with a 50 kHz sampling rate than for those with a 44.1 kHz sampling rate. But in about September of 1985, Apogee introduced aftermarket filters with linear phase response for non-oversampling 44.1 kHz decks. Oversampling 20-bit studio-grade converters were introduced by Apogee and others in 1993, and 24-bit units were introduced in 1997.

[4] Linear-tracking tonearms are expensive gimmicks designed to relieve rich people of their excess money, which I realized after designing a cheap one and wondering why the design hadn't been used previously. Even if a linear-tracking design manages to reduce tracking error, it won't sound significantly better, and it will have higher lateral inertia, which causes problems when playing records with off-center holes and/or warps. (There are tangential pivoting tonearms which reduce tracking error by rotating the headshell, but there's no concensus among audiophiles that they sound any better, and their prices are astronomical.) I also prefer pivoting arms because they allow a phono preamp to be placed near the pivot to minimize the length of the tonearm wiring. With modern op-amps, excellent phono preamps can be built into turntables. Well-Tempered Lab turntables, which are based on original thinking, use pivoting arms, although long ones in some cases. Unfortunately, they are considerably more expensive than what I would want to spend on a TT.

How Telarc put their 50 kHz recordings onto the CD-layer of SACDs

Rev 9/20/21

While looking into whether Telarc's early 50 kHz digital recordings have been digitally converted to CD-grade digital by using an asynchronous sample-rate converter (which became available in about 2006), I ran across a 2004 review of Telarc hybrid SACD-60634 (Saint-Saëns, Symphony No. 3 “Organ”, Eugene Ormandy, Philadelphia Orchestra) which was made by converting the 50 kHz master to DSD using a dCS 972 digital-format converter, then to analog, and then to CD-grade PCM using a custom Telarc ADC. The article indicates that they had tried using a sample-rate converter to convert directly from 50 kHz to 44.1 kHz, but that the results didn't sound very good due to problems with sample-rate converters at that time. The Wikipedia article on Soundstream provides detailed information on the digital recorder which Telarc used for making their 50 kHz recordings, which could be released as 50 kHz FLACs to minimize the number of sample-rate conversions which it would undergo in the process of being converted to analog.

I've also learned that someone developed a way to convert 48 kHz recordings to 44.1 kHz very early in the digital era, and that Decca released a lot of its 48 kHz recordings on a CD-collection known as The Decca Sound, which according to The Decca Sound by S. Andrea Sundaram, doesn't sound so great, although it's not clear exactly why.

So, there are probably other examples of early digital recordings, with sampling rates other than 44.1 kHz, which were transferred to CDs early in the CD-era. Such recordings were released as LPs, so there had to be decent DACs to convert them to analog, which could have been digitized with something like a JVC VP-900, an oversampling ADC with a 16/44.1 output, apparently introduced in 1982. So, it would have had 16-bit linearity and a linear phase characteristic, so that its recordings would have good low-level detail, good high-end detail, and good imaging, but it was expensive. In September of 1985, Apogee introduced its linear-phase aftermarket input filters for the typical early digital recorder, and recording engineers adopted them in droves as quickly as possible, so that most digital recorders soon had a clean high end and good imaging. But before then, a lot of digital recordings had poor detail and imaging.

However, mass-market CD players in general were lousy until at least 2010, due to the sound quality of low-cost audio-DAC chips. According to Benchmark's app note entitled A Look Inside the New ES9028PRO Converter Chip and the New DAC3 [November 14, 2016]:

"It has been a little over 7 years since ESS Technology introduced the revolutionary ES9018 audio D/A converter chip. This converter delivered a major improvement in audio conversion and, for 7 years, it has held its position as the highest performing audio D/A converter chip. But a new D/A chip has now claimed this top position. Curiously the successor did not come from a competing company; it came from ESS. On October 19, 2016, ESS Technology announced the all-new ES9028PRO 32-bit audio D/A converter. In our opinion, ESS is now two steps ahead of the competition!"

So, I gather that the 9018 Sabre DAC introduced in 2010 was the first really good audio-DAC chip, and it would have been too expensive at that time to put in mass-market players. But now there are many good inexpensive audio DAC-chips (although Benchmark is still partial to Sabre DACs), and Sabre DACs are appearing in low-cost players and DACs. I have a $100 2017 Nobsound Bluetooth 4.2 Lossless Player with a 9018 Sabre DAC, and it's amazing, although it's crude compared to Benchmark's DACs.

But due to piracy fears, the best versions of some albums are reserved for high-res streaming and LPs. According to various audio experts, including high-res expert Mark Waldrep, PhD (a.k.a. Dr. AIX), whose website is RealHD-audio.com, high-res recordings and LPs sound better than CDs because they're mixed and mastered better, and not because of the recording format. So, CDs supposedly could sound as good as high-res or LPs, if they were mixed and mastered as well, and the low-level detail could be kept out of the dither-region, where it is mixed with noise which is intended to mask the severe low-level distortion of 16-bit digital. There was a period in CD-history known as the "loudness wars," when CDs were recorded at the highest possible level because they would sell better, perhaps because high recording levels kept the low-level details out of the dither-region. Unfortunately, this approach required excessive compression and might have led to clipping.