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.

 

128x128Ag insider logo xs@2xshadorne
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If measurements were enough this hobby would be simple and we’d have healthier wallets. People who assume that everything we hear can be measured is a baseless assumption. Our hearing acuity is much more sensitive than current science can measure.

Not wanting to put down anyone’s system, but seems those who claim not to hear differences their DAC is not high enough quality, their audio chain is not transparent enough, or they simply haven’t tried demoing themselves.

Top DACs can be over 100k (DCS, Aries Caret, Wadax), with many top brands in the ~50k level (Total DAC, Lampizator, Linn).  Comparing DACs <5k level may show small/negligible sonic differences, but as you climb up a brand’s product line aka spend more for better engineering and parts, improvements can be heard

Like all components, it’s considered unwise to purchase a component beyond the ability of the audio chain (source, electronics ,speakers, cabling), money usually better spent elsewhere to lift sonics.  In other words, buying a quality DAC will not fix weaknesses in the audio chain, where the weakest link is the sonic bottleneck - the entire audio chain matters.

To me, NOS sounds more real than the upsampled, filtered options on my Pontus II. Listen to the echo decay of an organ in a cathedral. The decay with NOS sounds more realistic, 3D. Instruments have presence. 

@sudnh 

Do you mean the type of output (rca or balanced), output level, and if there is volume control? Or the final analog output filter design and buffer? Or the degree of separation of digital from analog side of things? 
 

For sure there’s a lot there to unpack

No..  I mean the analogue amp that is in every DAC. 
 

the implementation of the analogue amp also affects SQ. 

@sudnh 

Yes for sure the analog side of a DAC is important - the output with or without volume control is where the rubber hits the road! Absolutely!

@shadorne this is a great thread, thanks for initiating.  Have you tried FPGA DACs like those built by Chord?  I have had several of their DACs and have come to appreciate what they do - low listening fatigue, excellent timing, good detail and insane placement of instruments and sounds in the soundstage.  Wondering how you think your discussion applies to that design solution.

FWIW, I have tried software upsampling with JRiver V32 upstream of Chord DACs and find that while that can improve performance of cheaper delta sigma DACs, it actually reduces the performance of my Chord DACs which are wonderful at handling red book files all on their own.  JRiver may be significantly less capable than HQP which I haven’t tried.  Any thoughts on this appreciated.

kn

@knownothing 

I have no experience with Chord or JRiver. I don’t doubt your observations are correct - they seem to match what is observed by many others. Chord M-scaler is clearly a design intended to improve upsampling filters over the conventional approach. I read that Chord designer, Rob Watts, believes that highly accurate extremely long tap filters make an audible improvement over the conventional approach.