The Great DAC Mystery
This plethora of DAC’s phenomenon was such a mystery to me for 20 years. How can measurements be so incredible, yet many continue to prefer DACs that don’t measure so well. And almost everyone agrees they sound different (significantly in many cases). Why don’t the good ones sound the same. ASR are right in many ways - measured performance is important - but a pure focus on measured performance is completely wrong in my experience (using my ears). And here is my explanation of why!
Finally I believe I have stumbled upon a huge part of the problem with DAC technology. Of course it all stems from the inadequacy of measurements and even the technical instruments (audio precision) used to conduct those measurements - this is all at the root of why measurements are failing to be a reliable tool to select a DAC. There’s more though - if you read on please consider my reasoning and give my solutions a try - you may be surprised at the audible improvements that can be easily obtained.
There are a few things that hint at the problem of playing Redbook 44.1 source music:
1) R-2R DACs - why the resurgence?
2) Vinyl resurgence
3) The brick wall vs smooth, linear vs minimum phase debate: M-scaler, HQ player, FPGA XIlNIX proprietary programming, a plethora of filters.
4) HQplayer, PGGB and precursors like SACD - why is DSD still around and why do some people prefer it to PCM?
First let’s recognize that: All of these things can’t possibly be just coincidence!
So what is the underlying ROOT CAUSE:
Passband Ripple (‘equiripple’ to be precise)
1) All DAC’s are basically Sigma Delta DACs (which make up 99.99% apart from the recent handful but growing number of audiophiles with R-2R DAC’s). These Sigma Delta DACs ALL rely on upsampling to work - the final conversion is 1 bit or parallel 1 bit converters.
2) All upsampling DAC’s will take Redbook 44.1 (the vast amount of available music is in this format) and upsample (usually 8x initially but often higher) using short tap filters with low latency that have excellent specs but universally create a tiny but non-negligible passband sinusoidal ripple (it isn’t supposed to be audible).
MATH FACT: A sinusoidal ripple in the passband (what range of audio frequencies are presented to the listener) is equivalent to a pre and post-echo in the time domain (the signal you hear coming out the speakers)
The MANIFESTATION: Digital glare, harshness and a poor soundstage (the harshness is sometimes confused with accuracy - it is actually distortion - but not distortion that you can measure with an analyzer, as it is just like a reflection - it contains a reflection of the entire audio signal displaced in time at low amplitude ). Types of filters will have different forms of passband ripple - these lead to slight differences in the distortion (pre and post-echoes can occur at different times before and after the true audio signal - some time differences being more audible than others).
The SOLUTION:
There are three options
1)NOS with an R-2R DAC (can still suffer from aliasing which can create IMD in passband and the final filter can also create passband ripple)
2) upsample using a PC at such very high precision as to reduce passband ripple to inaudible levels (upsample can be to PCM or DSD but it might as a well be DSD as most DAC’s convert PCM to DSD anyway, only an R-2R DAC would be best fed upsampled PCM)
3) Vinyl - for the most part vinyl does not suffer from these issues at all but of course you get pops, cracks, surface noise, less channel separation, variability of pressing quality, and, if competing with digital; the need for very high end TT, phono-pre, cartridge, careful setup etc.
Anyway, please read carefully and think about the above with an open mind. Passband ripple is the elephant in the room that nobody talks about. Remember that very little if any testing has been done on our ability to hear pre-echoes however, anecdotally, all speaker builders recognize that a sharp baffle edge causes edge diffraction which is recognized as being audibly detrimental to the sound (and affects stereo imaging) Hence all the narrow speakers and exotic attempts to keep midrange and tweeter baffle width very small (think of all those countless big highly regarded audiophile three ways that are big on the bottom but narrow at the top)
It’s been a while, I thought I’d share this. No need to argue about this. I will offer clarifications but those who don’t get it or buy any of this will just miss an opportunity for better sound - I’d rather not argue with you. And, for those who will conflate pre-echo or post-echo with pre-ringing or post-ringing - I am NOT talking about ringing at all - the echoes I refer to are complete true echoes of the entire audio signal - equivalent to and analogous to a reflection off a wall.
@baylinor nice one (Jack Handey would approve) |
@shadorne, your item 3 has always struck me as a highly significant issue. When the Redbook standard was announced back in the late '70s, I heard about it from a physics teacher friend, and my reaction was that a 44.1 kHz sampling frequency was too low, probably be a factor of 2 or more. We agreed that virtually any filter would introduce phase shifts at frequencies well below 20 kHz, which would likely be audible as degraded transients. And I heard that from the dawn of the CD age. There's another, related issue. The engineers who invented the A to D and D to A processes relied on the Shannon-Nyquist theorem to take the position that the DAC output would be indistinguishable from the ADC input. The problem is that the theorem, as it is usually stated, states that the sample rate must be at least twice the bandwidth of the signal. However, this is an incomplete statement of the theorem--in reality, properly stated it says the sample rate applied to a continuous function must be at least twice the bandwidth of the signal. And music is not continuous, so you can't rely on Shannon-Nyquist; there will always be some variance, sometimes audible, sometimes not. But in practice, both the application of a higher filter cutoff and a the concomitant higher sampling frequency both reduce the variance between input and output. |
@corente plus1 You said before I did.😁
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@shadorne Thank you for your post, very interesting! Theoretically speaking, would oversampling all digital signals to DSD256 and feeding that to a non-oversampling, DSD-only DAC (or at least a DAC with a discrete DSD stage) provide a workaround to the issues you are describing?
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The subject matter is waaay beyond my comprehension but to all those saying “just go with what sounds good to you” I have these thoughts. As a longtime admirer of David Hafler, Edgar Villchur, and Peter Pritchard, engineers who made great sounding products that were affordable by the many music lovers, not only the few with high incomes, their goal was to discover what was musically relevant in circuit or material science, and apply clever methods to make products that excelled without needless cost. Hence the PAS-3, Mark lll, AR XA, 3a, and the XLM, Sonus Blue Label, etc. If “passband ripple” is reliably identified as a key source of audible signal degradation, and an economical way to rid it from the D/A process can be found, then we all benefit, at all price levels…at the Topping ChiFi and the dCS budgets alike. That would be progress. |
Interesting, way beyond my comprehension except for the understanding DAC’s aren’t able to convert, output signals that measure perfectly. I’ve had traditional DAC’s, R2R, FPAG, DAC’s with a Tube stage. Currently have a PS Audio that is a FPAG, doesn’t have any of the off the shelf chips and converts all signals to DSD and outputs them in DSD form. Believe the measurements aren’t that great on the DAC. Designer talks about designing the DAC around being able to tune the DAC to shape the frequencies in a way that sounds best to our ears. Interesting that following that format produced a DAC that for some measures poorly but to many sounds great. OP, how does your theory, principles you reference apply to FPAG DAC’s? |
Long post. You appear find satisfaction and frustration in contemplating, philosophizing, and pontificating on audio engineering problems and solutions way beyond my level of comprehension. My recommendation is to relax using whatever meditative or chemical means you wish (good bourbon or cognac for me) and leave these issues to the audio design engineers. Get back to the basics of this hobby, the enjoyment of music and the pursuit of the most natural sounding system within your budget. Listen to live music, both acoustic and amplified. I always calibrate at Carnegie, the MET, the NYC Ballet, or jazz and rock bars in the Village. Establish listening principles to develop your perception of live music. Develop good auditioning methods (much is published on this). Forget about measurements (except for determining system compatibility such as impedance matching) and design. Find a good dealer that is knowledgeable to assist in system compatibility and set up. Use your ear-brain connection to determine what equipment meets your perception of good SQ. Use reviews only for preliminary research to find equipment that may meet your goals and perceptions. Build the best system within you budget that sounds right to you. Chill and enjoy the music. |
Late last night was one of those special times that all audiophiles know what I'm talking about. Whether it was low usage on the power grid, atmospheric pressure or special alignment of the stars, the music was just flowing and every audio aspect was nirvanic perfection. Record after record sounded better than the one before and knowing that I would soon have to call it a night and that it would be quite awhile before these conditions would come again, I went deep into my archives and pulled out some of my favorite measurements and immersed myself in numeric bliss. |
@shadorne Based on my very limited knowledge of D/A conversion, I can’t decide if you’re a genius or a kook. I do believe that the quality of the DC power supply and the nature/quality of the analog output section both have a significant influence on the quality of the sound delivered by any particular external or internal DAC. If external, then the quality/nature of the interconnects (cables) and interfaces (USB, SPDIF [in both optical and coaxial {RCA and BNC} renditions], as well as AES/EBU) must be considered. As an end user, not an engineer (as you seem to be), I can only judge the quality of the conversion by (1) what I hear (2) in my system. I have to get my critical listening ears on to hear significant differences between the dual Burr-Brown, AKM, and SABRE DAC chips in my current inventory. As soon as I start listening to the music, any distortions I might have heard disappear: I am transported. |
Just a comment on measurements. I heard a one off custom set of speakers that were wonderful. The designer was very technical. And always looking for correlation between any measurements and what's heard I asked him how he used measurements in his design process. His answer was to eliminate errors, things that are obviously wrong as measurements show. The rest he did with his ears over long time periods. |
I listened to several DACs between $1500 and $5500 /$6k and left with the LAB 12 DAC1 Ref. To me it was exactly what I was looking for. It was a little more than my budget but I wanted to avoid a lateral move from my RME. That's not to say the Merason Freot wasn't good, it was. The Bricasti was excellent too but at $3200 the LAB12 was more my budget. After 2 years I still really like this DAC. It is so nicely balanced in MY system, complements my system perfectly. It would take an in home audition that blows me away to even think about changing out that DAC.
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@roxy54 +1 |
I'd guess it's because ears, rooms, brains, and gear are not the same as mics and measuring equipment, and isolated chambers. Unless you live where the measurements were taken, they will likely bear little resemblance to what you actually hear in your environment. That's why I pay little or no attention to measurements; nothing to do with me. |