Measure twice, cut once. That's my deep thought about it.
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.
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- 60 posts total
@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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@corente plus1 You said before I did.😁
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@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. |
- 60 posts total