2. digital
  3. 191748-268

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.

 

shadorne

Showing 6 responses by asctim

asctim
507 posts
 

An interesting article about DAC filters and ripple:

https://addictedtoaudio.com.au/blogs/how-to/how-to-pick-the-best-filter-setting-for-your-dac

I remember reading a Stereophile article way back in the old days. One of the reviewers had got into his possession a DAC of some sort that would let him write and apply his own filters. He was pretty sure that ringing filters were the culprit for what he was hearing that he didn’t like about digital. His conclusion after writing and listening to some horribly ripply filters was... filters weren’t the problem. He couldn’t make the DACs sound any better or worse to his ears by eliminating ripple.

I bought a NOS, filterless dac with an old chip because I wanted to hear it for myself. I couldn’t hear any improvement like I was expecting from that either. I think most of what people are hearing and referring to as things like microdynamics and timbre are happening on much longer time scales than any problems with timing that digital is introducing, or pretty much any electronic gear such as amps or pre-amps. However, anything that alters frequency response even just a little bit can result in all sorts of unintuitive perceptions about timing, I’ve tested my hearing on timing issues, creating signals that I thought would audibly reveal timing issues with slower bitrates compared to higher. Measurements showed the higher bitrate effect was actually coming through the speaker, but I definitely couldn’t hear the difference. That’s when I realized I was making timing effects that occur faster than 1/20,000 of a second. I can’t hear that high, so I can’t hear the timing that high either.

asctim
507 posts
 

I did some reading on the subject. I read the echo is under 1ms offset and -60 dB or more below the main signal. That’s really quiet, and not a lot of timing offset. The thing to show is if under any conditions anyone can pick up on that, maybe not even hearing it directly, but sensing it somehow changing their perception of the sound. Perhaps some people can. I suspect I can’t.

I’ve tried mixing echoes of various types in to recordings to see what effects it might create. I quit hearing any notable change by -35 dB. Maybe even -30 or -25, depending on the type of echo and the timing.

If you want to hear something that really stands out, use a pitch shifter to slightly de-tune the echo. That’s a really bad sounding effect. Even that one disappears for me long before -60 dB.

asctim
507 posts
 

I can't find any mention anywhere of oversampling causing pre and post reflections. Can you site some references?

asctim
507 posts
 

https://src.infinitewave.ca/

Here's an interesting web app. to compare sample rate converters. I've definitely heard the weirdness that happens sometimes when digital goes off the tracks. If you play a sweep you hear very distinct echoes when this sort of problem is happening. It's like an effects generator.

When it's really bad you can definitely hear it. At -67 dB maybe some people can, but I doubt it's one of the main causes of audiophile dissatisfaction with digital

asctim
507 posts
 

This is a mathematical certainty - no hand waving at all.

Agree. I'm convinced it happens. I'm not convinced it's typically an audible problem. I just did an experiment with 0.7 ms pre and post echo loud enough to be easily heard. At -10 dB the effect is loud and clear. Besides being obviously louder, it also sounds fuller with more midbass energy and dramatically reduced treble brightness. Definitely not harsh and gritty. Using REW to combine artificially generated measurements to get one with a -10 dB pre and post echo at .7 ms shows comb filtering starting at about 200 Hz with minus 13 dB nulls occurring all the way up, leading to an average sound level above 200 Hz of roughly -6.5 dB compared to below 200 Hz. That explains the reduced brightness.

At -67 dB the comb filter nulls are about .015 dB. To call that subtle would be a vast understatement. 

asctim
507 posts
 

The 0.7 ms time delay correlating to inter-aural distance is an interesting point. Listening to the effect at -10 dB reminds me of what I hear in the phantom center on a typical stereo listening triangle. It's darker sounding compared to sounds panned to the sides. I've found that using a wider listening triangle works better for me. The first null is pushed down to about 700Hz, which is similar to the results of my experiment. But the benefit I think is better head shadowing. So the phantom center overall sounds brighter and clearer to me with a very wide speaker spacing.

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