PCM v DSD Comparison: 16/44.1, 24/96, 24/192, 64x DSD, 128x DSD

There's a lot of buzz about DSD, talk of CD's demise, and music offered in various PCM formats from CD-quality up to 24/192.  What's a person who cares about the quality of their music reproduction experience to do?  And the answer is, listen.  I got the idea to listen to various files in different resolutions and formats after receiving some comments on my Mytek Stereo192-DSD DAC review asking about double rate DSD or 128x DSD.  Namely, is 128x DSD better than 64x DSD.  It sure sounds better, mathematically!  But the only way I know to figure out if one thing sounds better than some other thing is to listen.  So that's what I set about doing. 

Let's get a few things straight before we dive into listening.  First off, this comparison is necessarily limited by a number of factors including the fact that I'm using one DAC, the Mytek Stereo192-DSD DAC (see my review here), 3 tracks each in 16/44.1, 24/96, 24/192, 64x DSD (2.8MHz) and 128x DSD (5.6MHz) all sourced from the same master tapes and all converted using the same process (see below).  I do not intend to draw any sweeping conclusions since I'm aware of the limitations that the environment I'm listening in necessarily imposes.  That said, I believe its fairly safe to extrapolate somewhat but I will try to keep my reaches within an arm's length. 

Let me fill out the rest of my setup for this listening session: a PC running Foobar2000 connected to the Mytek Stereo192-DSD DAC with an Audioquest Carbon USB cable, a pair of Kimber Kable Select KS 1126 Balanced cables to the Pass INT-30A, and out to my DeVore Fidelity The Nines.  I'm using a PC because at present the Mytek DAC only supports 128x DSD via ASIO. 

Credit Where Credit Is More Than Due
I owe a huge thank you to Bruce Brown of Puget Sound Studios (you can read our Q&A here) for providing me with the music under scrutiny.  Bruce was kind and generous enough to agree to create these files for the sole purpose of this listening experiment and I want to thank him for his time and for providing some great music to listen to over and over again. 

Here's a word from Bruce on the process he used to generate the various file formats:

These were original master tapes that I had of these tracks.  I made one tape dub of these 3 tracks that would facilitate the test more easily.  The tapes were recorded to RMGI SM900 tape via a Studer A80RC MKII that has been greatly modified.  This was also the playback machine.  I created test tones so the tracks would have equal volume and the tape was played 5 times into each sample rate.  I started off the test at 16/44.1 into the newer Korg MR2000sBLK using Mogami Gold 1 meter balanced interconnects.  No EQ, Compression or Gain was used, just a straight transfer.  The files were then transfered and labeled to an external hard drive and sent to you via zipped ftp. 

Thanks Bruce!  I'd also like to thank Mytek for the very extended loan of their DAC for the purposes of this comparison. 

More Details
And here's the details of the Foobar2000 setup for 128x DSD playback over USB (this is detailed in Mytek's setup PDF):

Upgraded the Mytek to firmware to 1.7.1 (if you experience static along with your DSD music, make sure you've upgraded to this latest firmware and it'll go away)

Let the listening begin! 
There were three sample tracks provided [to avoid copyright issues the tracks names will remain private].  All PCM tracks were provided as WAV files and DSD as .DFF and again we have a 16/44.1, 24/96, 24/192, 64x DSD and a 128x DSD copy of each track.  I loaded up all tracks into Foobar2000 and spent time listening.  I listened to all tracks all the way through a number of times and then went back and focused on certain parts of tracks that emphasized or highlighted the differences I heard.  And I will say up front that the differences between CD-quality and DSD make me want to put that word quality attached to CD in quotes. 

16-bit/44.1kHz
Let's start on the positive side - music sounds punchy and bold and is clearly well recorded.  There is also a sense of dynamic edginess, a hardness that creeps into the sound at dynamic peaks as if you're not hearing the full sound.  This holds for each track and lends the music an overall sense of harshness as if it was recorded in too small a space.  There's also an unnatural aspect to decay and reverb where they sound stunted reducing both a clear sense of the space of the recording as well as the full natural voice of an instrument or vocal or a finger snap.  There's also an emphasis on transient attack that shifts the music's tonal center away from body. 

24-bit/96kHz
Everything opens up as if there's more space in and around the recording.  Decays sound more natural, musical images in space take on a more solid and fuller feel.  Dynamics have a greater sense of ease, tone colors are richer and overall music sounds more natural and relaxed.  On the loudest passages which are most prevalent on Sample #2, there is still some sense of hitting a wall as if some of the sound has been cut off. 

24-bit/192kHz
The size of the recorded space is once again larger and much more natural sounding, even compared to the 24/96 versions.  There's also a better sense of micro-detail or hearing exactly what the musicians are doing.  A more intimate picture.  This lends the music more drama, more impact because there's more variety to the sounds.  Dynamics also appear to have a greater swing from soft to loud and there's an overall sense of ease that is not present in the previous versions.  Upper frequencies take on a sweeter sound, cymbals sparkle, and horns sound more fleshed out and with less glare. 

64x DSD
Holy crap!  The musicians just relocated into real space.  Finger snaps are 3D as compared to the PCM versions.  Actually every aspect of the music is more dimensional.  Dynamics are astoundingly natural-sounding and there's an overall ease to the presentation that translates into an almost uncanny sense of place.  Instruments take on a more complex tonal palette and you can hear into the performance much more.  Harmonica sounds like a harmonica as opposed to a piece of one.  The CD-"quality" version sounds relatively dull and lifeless as if we're missing out on a ton of detail and subtle nuance. 

128x DSD
Space is the place.  Dynamic swing sounds unrestrained and fluid.  There's absolutely no sense of harshness, edginess, or harmonic foreshortening.  Music is rich and full.  The scale of the recorded space is rock solid and stable and music emanates from this space in a completely natural way.  CD-"quality" sounds like a cardboard cutout in comparison.  There's a lot of "space" in the Sample #1 recording and the opening vocals resound in that space.  With the double-rate DSD you can hear the size of this space whereas with the CD-"quality" version it sounds as if the singer's voice is hitting a wall.  There's no depth, no complexity to this reverb tail with 16/44.1.  With each step up the PCM bit/sample rate ladder and with DSD this tail gets more and more fleshed out, solidifying a sense of time (distance) and place. 

Conclusions
The differences I noted between 16/44.1 and DSD are dramatic and easy to identify.  Even though I was dealing with very good-sounding recordings, you could even say the CD-"quality" versions sounded good, when compared to higher rate PCM formats and DSD you get an increased sense of dynamic ease, harmonic complexity, micro detail, and a better sense of the recorded space (and time).  A more natural-sounding presentation.  The differences between the 24/96 versions and the 24/192 versions were not as significant but differences were there to be heard none-the-less.  The same held for the two DSD versions so the most obvious jump was from 16/44.1 to 24/192 and DSD. 

I admit that this could all be due to the Korg MR2000sBLK simply being better at converting to DSD and higher rate PCM as opposed to 16/44.1 or the Mytek DAC's ability to reproduce DSD over PCM.  But I've also heard similar improvements with other recordings albeit under less controlled circumstances.  My feeling is the differences noted between CD-"quality" and 24/192 and DSD are so marked as to suggest that the medium goes a long way in conveying the message. 

Sample #2 opens with the band's percussionists playing soft to loud to louder still.  In the CD-"quality" version this sounds flat and stunted as if it was recorded in a room just large enough to fit everyone, nothing has room enough to breath.  When you listen to the same track in 24/192 or DSD you easily realize the musicians are mimicking the sound of a train approaching going from far to near, to nearer still as opposed to just going from soft to loud.  And you realize this because you can hear into the recorded space and the music comes from a deeper, quieter place.  This train reference is central to the message of this song, "...they curse this train that brought them to Johannesburg", so there's more drama, more emotional impact and the higher you go up the PCM ladder the more of this drama you get and you really jump right onto the tracks with DSD. 

"How can I be sure..."
I think Bruce Brown was teasing by picking the song for Sample #1, each time I played it I wondered why some doubt was creeping in (kidding).  I figured it was worth putting my findings to a test so I put all tracks on random play, turned off my monitor, took off my glasses (which is the same as being blind) and let the music play.  Out of 8 random trials (this was all I had the patience for), I was incorrect once thinking that I was listening to was 24/96 when it was in fact 24/192 (this was my first trial and I rushed my decision).  All other selections were correct. 

But I'd say I got lucky with some picks.  The differences between 24/96 and 24/192, for example, are not huge but there are differences if we listen closely.  It also helps to listen to complete songs as opposed to switching back and forth between snippets.  I do feel confident that I could identify the 16/44.1 version each time as well as identify PCM v DSD but even here Sample #2 and Sample #3 were easier to determine as compared to Sample #1.  Then again, if you asked me to do this again in a few months using your computer's speakers all bets are off. 

Getting Engaged
This was an interesting and informative comparison and I hope I conveyed a sense of the differences I heard.  More importantly I'd like to stress how these differences can affect the way we perceive music and how they can impact the meaning of a song as opposed to just a focus on sound and sound effects.  For me its clear that 24/96, 24/192 and DSD are superior to 16/44.1 in many meaningful ways and I've come to this conclusion based on listening to more than today's 3 tracks in different system settings.  But the ability to listen to these various samples all generated from master tape using the same equipment and methods, thanks again to Bruce Brown, has helped to solidify this belief. 

Where does this leave us?  Well, there's the pesky issue of the quality of the original recording which obviously trumps all of the above.  Then there are all of the variables, some of which we've touched on here, which makes drawing any firm conclusions potentially misleading.  But to my mind when speaking in generalities why not go for ideals?  Sure CD-"quality" can sound really good but higher bit/sample rate PCM and DSD can sound better.  How much better will come down to the quality of the recording, the quality of the transfer, the playback chain, and your listening habits and preferences.  Better still, if you value music first, all of this takes its rightful place in line. 

You have to wonder how these PCM tracks would fair if they were converted (Torn?  Un-ripped?  Pired?)  using an Ayre QA9 or something similar.  Perhaps the greatest difference between PCM and DSD is the filters used - or not used - in the process.  Better filters might make all the difference.  Or, maybe they do but there's more to the story as well. 

Are we back to buying really great vinyl, playing it on a really great turntable/arm/cartridge/phono preamp and piring it all with a great A/D just so that we can get really high quality digital playback?  Think of the hobby hours one could devote to just that.  Nobody would have a minute to read Audiostream. 

Yikes!  I certainly didn't see that coming and I'm sorry you found it so. 

I see all of this as a positive since we're talking about an obvious improvement over CD-quality.  And not everything is available on LP.  And digital recordings, especially HD digital recordings (24-bit and DSD), delivered in their native format sound marvelous.  Lovely.  And tape to HD or DSD can also sound marvelous.  Lovely, even. 

I have some doubts that most of the fine record companies we have all grown to know and love (you know the exceptions) will ever embrace releasing high resolution files.  And, if they do, some modern creative genius - in other words, not Bruce Brown - will provide his or her own brilliant remix in the tradition of those Perfect Sound Forever 12 bit CDs from early on. 

Since I aquired a Mytek DSD DAC, I also acquired about 20 albums in DSD.  Most are classic rock and jazz that were transferred to DSD (SACD).  A few (classical) are native DSD recordings. 

The native DSD recordings sound amazing.  Analogue like and realistic, but with all the advantages of digital.  NONE of the negative "digital" sounds often complained about. 

Many of the analogue tape to DSD albums also sound great.  Certainly more lifelike, realistic, and analogue sounding than their CD counterparts.  Not as good as the native DSD, but really good, and not "digital" sounding.  It's clear to me that analogue direct to DSD is the best way to digitize older material. 

I also have lots of PCM hi-res from HDTRACKS.  Mostly tape to hi-res digital.  Some are analogue to DSD to PCM, others are straight to PCM.  Again, I find the hi-res just sounds more relaxed and realistic than Redbook, and definitely gives a better sense of space and instrument location.  The exceptions are when the hi-res is fiddled with too much after the conversion, and volume compressed. 

So I don't care what all the "objectivists" say, or all the engineers that can explain to me why hi-res "can't" sound better than redbook.  I just enjoy listening to it more, and it gets my foot tapping.  That's all that really matters. 

I just wish there was a lot more classic rock and jazz released in DSD, or at least in 96K. To really build a library that would fit my tastes, I'd need to locate the right PS3 and start buying and ripping SACDs.  Something I really don't want to do.  Unfortunately, there doesn't seem to be much chance of that older stuff being released that way, as the big labels don't seem to want to allow true "master tape" quality files out in the marketplace.  Hopefully they will start/continue releasing new material in some form of hi-res. 

I think we can take these findings and use our computers to optimize our redbook.  I have personally found that using Sox with min phase filter, dither, and allow aliasing when upsampling 16/44.1 to 24/96, gives better, smoother more realistic sound that native redbook. 

So, yes, these findings are helpful I believe to optimize our music libraries, whatever the resolution. 

Better to go to 88.2 with SoX and 44.1 KHz input files.  Less rounding error due to integer conversion and the internal word length used by SoX. At least that is my experience.  No dither needed when adding more bits to the output file.  YMMV, etc. 

I tried going in multiples.  I still felt 96khz was 'better'.  Its funny, because I was just listening to an HDCD of Buffalo Springfield.  It was ripped to 24bit/44.1 with dbpoweramp.  I had both 'native' and upsampled Sox 24/96 in one folder.  Listening to each song back to back.  Very little difference between the two.  I ended up deceiding on the 24/96 due to a very intangible improvement in 'fluidity' that I could identify. 

Unfortunately I packed up the Mytek DAC for return.  I may be able to get to this in a few weeks with another DSD-ready DAC but we're getting into even more variables including what software we're using for the PCM > DSD conversion. 

And in a similar vein, some players, like the HQPlayer from Signalyst, allow you to upsample everything to 128 x DSD on the fly. 

I am so envious that you have all the different file formats derived from the same master tapes and mastering equipment!  I have always wanted to do such a comparison but have never able to find the availability of such files that are fit for the purpose.  Do you think Bruce will mind if you share the files for the sake of personal observation and experimentation?  I would love to experience what you have experienced personally as well! 

I have the Mytek too and the most recent experiment I had conducted was to upsample all 44.1/88.2/176.4kHz files to DSD128 in real-time to stream over to the Mytek for decoding with Type D SDM (the lastest foobar SACD plugin (v0.6.1) by Max has an experimental provision for 44.1kHz based materials to be upsampled to either DSD64 or DSD128 on-the-fly).  Personally, I feel all the upsampled material has been given a new lease of life.  i.e. they end up sounding much smoother and less fatiguing to listen to, more analogue-ish if you may.  Perhaps you would like to give it a spin as well!  I guess the concept stems from the way in which Playback Designs and EMM Labs upsamples all PCM material that are fed into their DACs to DSD128 via hardware implementation. 

I've posted elsewhere about this, but here is a perfect chance for those who have DSD-capable DACs (which they like in PCM, too, of course) to test the issue.  There is a DVD-A/SACD 2 disc set by a small label called Ludomentis, weherein pianist Massimo Gon plays Liszt's Grand Etudes to two different signal paths, where one is recorded directly to 24/192 and the other directly to DSD.  To quote from SA-CD.net "They set up a 5.0 array in a near single-point source configuration with Shoeps microphones at front row distance from the piano, recording direct to hard disk in DSD (and PCM for the DVD-A disc).  Editing was in DSD or high-res PCM realms as appropriate." 

What's nice is that this recording can also be compared in mch 5.0 DSD vs 5.0 PCM too.  On my Mytek, and on the Exasound E20 I have currently inhouse for demo (never tested this disc set on the Sonore eXD), the differences are consistent and yet not terribly important.  I'd live with either (and I am a hufge DSD fanboy with 850+ SACDs ripped, etc) but ultimately like the attack on DSD better.  With my main rig DAC, the Meitner MA-1, the DSD is clearly preferable by a larger margin, with much better piano attacks and decay, and a better sense of the venue.  I take this to be DAC specific, as my Meitner uses 1 bit chips and upsamples eevrything to DSD128. 

And it also reminds us that the totality of the chain matters.  Regarding the first commenter's lament, yes, sometimes the best sound will come from investing the time and cash into a great analog system and making your own digital copies. 

A lot of what's been sold as hires conversions of old material is presented either with squashed dynamics, "creative EQ" applied or was possibly done with an ADC that's no better than what you can buy today as a consumer/prosumer.  JA's recent QA-9 review in Stereophile was a real eye-opener and hopefully a more complete opening salvo for others to follow. 

That said, digitizing LPs is a hobby within a hobby within a hobby.  The good news is that often the best source material is a cheap used, orig.  LP and not an expensive reissue. 

In my experience for PCM, 24-bit, any resolution, is clearly better than 16-bit.  As to what sampling rate matters to you I think that's as much a function of how well your particular ADC or DAC performs at a given sample rate as the sample rate itself might influence the outcome. 

I'll also add that the apparent reduced number of common choices for DSD formats (and yes I know there are a few more than were mentioned here) are a practical advantage when discussing or choosing "what's best."  The plethora of sample rates for PCM strikes me as a paradox-of-choice problem overall, and an unfortunate distraction usually. 

The original recording quality is definitely important, but IMO, there is something fundamentally different and more 'real' to any high res.  data when compared to redbook.  Its a certain 'connectedness'.  Its just more real.  I can appreciate more what the vinylphiles appreciate when I up the sample rate. 

Has a pretty interesting selelction of 24/192 - some classic rock and jazz.  Mostly converted from DSD (there seems to be more SACD being released in Japan than in the West).  Interesting they don't offer straight DSD downloads. 

I have good experience with analogue converted to DSD; how does conversion to 24/192 affect SQ?  Is it barely audible, a minor change for the worse but still really good, or a serious reduction in quality? 

Onkyo e-music says they are going to open a US store - will be interesting to see what titles they are licensed to sell, and in what format. 

When i see an article like this where the subjective evaluation between 16/44 and DSD128 appears so MASSIVE written by someone who tries to control the variables, I have to ask one simple question...

Why not upsample the 16/44 to DSD128 and ABX the two files, then post the results?  The upsampled 16/44 should be aweful given the extra PCM -> DSD step, right? 

If the writer cannot put his/her hearing and subjective certainty on the line for such a simple test, then this article is just more BS to tout DSD for whatever reason... Addendum: even better how about posting samples (30 seconds?)  of both DSD128 for folks with DSD DACs to try for themselves (if permission is granted). 

Then why don't you try something so simple... I think it's just intellectually honest given what COULD BE hyperbolic subjectivity. 

Look, I do appreciate the article - good entertainment for the hobby.  But when big claims are made, why not dig deeper?  To be able to convincingly show an objective difference would strengthen the case massively for many to strongly look at getting into DSD. 

An ABX test, as Archimago suggests, is listening.  It is simply a more intellectually honest form of listening than the one you undertook here. 

Here's the thing: I want to agree with you.  I have DSD recordings that, to me, sound better than their red-book counterparts, and HD downloads that, to me, sound better than an equivalent ripped CD.  But I also think it's important for me to know whether I'm fooling myself or not. 

Granted, you're not obligated to do anything you don't want to do.  But to claim that one sounds better than the other without taking the step of eliminating the variable of knowing what you're listening to means that your opinion is just that, and it carries no more authority than mine, a first-time poster. 

The fact is, more and more science is lining up against the audiophile "opinion" that red-book audio is insufficient compared to 24/96, DSD, or anything else.  If you're not willing to engage the science by walking into the ABX lion's den, then the rest of us have every right to question that. 

I'm very comfortable with the fact that I may be fooling myself.  It's my brain, I can do what I want with it.  You don't seem willing to admit that might be the case. 

The fact is, more and more science is lining up against the audiophile "opinion" that red-book audio is insufficient compared to 24/96, DSD, or anything else. 

I'd be interested in reading some references. 

If you're not willing to engage the science by walking into the ABX lion's den, then the rest of us have every right to question that. 

Of course you can question my findings and I'm perfectly OK with that.  Ideally, people will be curious enough to listen for themselves which, imo, is the only way to determine one's listening preferences. 

As for my own ABX tests, I'd be happy to do so, except for one problem: I can't reliably gather proper source material like you can.  Even for my HD downloads, I can't guarantee that they were produced from the same master as the CD rip I'd be comparing them to, so any test result would be suspect.  Beyond that, I can't compare DSD to PCM at all because I don't have any source for the former beyond my SACDs, which I can't rip.  Finally, not all of us are blessed to have access to as quiet a room, or as high-quality equipment, as would be necessary to do a test like this in a scientifically-valid way. 

That's why I think it's important that folks like you, who have the means to do so and are making the claims, to do these tests. 

The author cites 2 papers - Dan Lavry's white paper "Sampling Theory For Digital Audio" from 2004 which talks about an optimal sample rate of 60KHz and "Coding High Quality Digital Audio" by Bob Stuart of Meridian.  Here's a quote from that paper's conclusion, "The CD channel with 44.1kHz 16-bit coding (even with noise shaping to extend the resolution) is inadequate...Even 48kHz sampling is not quite high enough..."

So there is clearly some argument to be made on a technical basis to refute Monty's position just by reading the papers Monty cites to support his own argument that is summed up here, "Empirical evidence from listening tests backs up the assertion that 44.1kHz/16 bit provides highest-possible fidelity playback." 

In terms of my good fortune, I find myself in this position largely because I value the experience of listening to music on a hi-fi.  The enjoyment of listening to music on a hi-fi is not, imo of course, reducible or adequately represented by any white paper or ABX test.  The true measure of listening to music on a hi-fi is time.  Time spent listening and ideally enjoying music.  So if someone like Monty prefers to listen to lossy encoded files or limit his experiences to 16/44.1 who am I to tell him to do otherwise? 

I prefer to listen to what I find to be the most musically engaging experience I can get and I have found that higher resolution PCM and DSD formats (and LPs) typically provide a more musically engaging experience. 

2.  Those two papers you mention here also contradict you with respect to there being any difference between 24/96 and 24/192, and Lavry's directly argues against 192kHz.  So I wouldn't lean on them too heavily. 

I also read your original response to the xiph paper, and besides the obligatory snarkiness (hey, that's why they invented the internet, right?),  I didn't see anything you wrote to actually refute anything he wrote.  And with respect to Neil Young, I love the man, I love his music, and I believe that he believes that he hears a difference.  But he has also been playing electric guitar on stage at ear-shredding volumes for four decades or so, so I don't know whether I would lean on his opinion on the relative audio quality of 16/44.1 vs 24/192.  Honestly, I think I'd take your opinion over his. :-) 

You say you "appreciate his... advocacy of double-blind and ABX testing," but not enough to actually do it.  Considering that there are some reasonably smart people arguing against your position from a scientific standpoint, I'd think you'd want to take a crack at it.  Why not go for it? 

Because they have something to sell.  And they don't really "argue" against it in any scientific way, they put out marketing fluff and lean on consumer assumptions. 

In the context of "enjoyment of listening to music," I agree completely that double-blind and ABX testing do not apply.  But that is not the context of this article.  This article is about which sounds better, and in that context, double-blind and ABX testing do apply, more than your opinion does. 

I'm beginning to think the emperor is feeling more of a breeze than he's letting on. 

So all of the engineers that design DAC chips, for example, capable of handling 24/192 and DSD actually know that there's no point to it other than hoping that some DAC manufacturers who also know that there's no benefit from higher sample rates will make DACs using these chips so they can sell some stuff that they know sounds worse than 16/44.1.  And of course the recording engineers who offer HD music are all in on this conspiracy as well and the only reason people believe they hear an improvement with HD music is because they've been told they will in "marketing fluff" when in fact it sounds worse than 16/44.1.  And you believe this based on a few things you've read that contradict what you yourself have experienced....

Here's the thing: I want to agree with you.  I have DSD recordings that, to me, sound better than their red-book counterparts, and HD downloads that, to me, sound better than an equivalent ripped CD.  But I also think it's important for me to know whether I'm fooling myself or not. 

Your emperor appears to be less well dressed than mine. 

btw - there are free downloads available from SoundKeeper Recordings that offer files in various bit/sample rates (16/44.1, 24/96, and 24/192) all sourced from the same master so you can perform your tests yourself. 

I didn't say they knew there was no point.  There are entire industries where the people who are selling something geniunely believe they are solving a problem and helping their customers, but it's still based on misinformation and assumptions.  I'm not putting this in that category per se, but given the papers we've referred to here, I'm keeping an open mind about certain aspects of it. 

And yes, the things I've read have made me question what I have experienced.  How could it not?  The science in them makes sense to me, and there isn't really any contradictory evidence out there.  Do you really believe you are immune from confirmation bias?  It's part of being human.  The only way to prove it is to take your foreknowledge out of the equation.  The amazing thing to me is that you don't seem to be the least bit curious about it. 

It's clear that you're going to use every rhetorical method you can think of to avoid an ABX test, and still be able to maintain that you can tell the difference between 24/96 and 24/192.  There is science that says you can't, and you're unwilling to prove you can in any scientifically meaningful way, so.... It's a shame, because after stumbling onto this site not too long ago, it seemed like a place where I could learn a few things.  But you have better things to do than chase one reader. 

I did in fact listen to a random sampling as I discussed and was able to correctly pick 7 out of 8 times. 

The issue I have with ABX tests is they do not prove anything significant in this case, imo.  For example, let's say I correctly pick DSD v PCM every time (which I did without knowing what file was playing) and 24/96 versus 24/192 the majority of the time (which I did).  And let's say we can also easily show how these files differ from one another in terms of their underlying data (which we can).  The most important question remains - what are the perceived differences and are they worth it?  And the only way anyone can know the answer to this question is to listen because of the number of variables involved including music choice, system, room, and personal preference.  That's because we're talking about listening preferences in music, not some objective criteria that is either met or not. 

But the real question to address your point is, why didn't my - for all intents and purposes blind test - suffice for you? 

It's a shame, because after stumbling onto this site not too long ago, it seemed like a place where I could learn a few things. 

If the only way you feel can learn something is through reading about ABX tests, then you will have to look elsewhere and I agree that is a shame. 

This is a friggin hobby.  For enjoyment.  As in smiles, fun, good feelings, happiness, to name a few possible descriptive terms.  I am constantly amazed how many people don't seem to be having fun.  But, to each their own. 

From the engineering, or scientific side if you like, the sampling rate and bit depth for mixed signal fidelity are just a couple (albeit very important) considerations of many.  This is true not just for audio reproduction, by the way. 

I have been talking about using various software upsampling techniques, filters etc for a couple of years now.  Its an easy and cheap (free) thing to do, and anyone can try it (foobar, Sox etc).  Anyone can take a redbook file and upsample it, play with the filters, dither, aliasing etc and see what happens. 

I have found, that in the tests that I have done, I like the upsampled files better than the native 16/44.1.  Not by a ton, grant you.  I can listen to them back to back.  Back and forth, over and over.  It would clearly follow, IMO, that a remaster of an analog tape to these various sample rates/resolutions would sound different, with an advantage to the higher rates.  This just makes sense to me. 

What I would say to the doubters: do some work of your own.  Download foobar, learn how to use it, add in some open source SRC plug ins, and see what you get.  Discard it if you dont like it, but more importantly, tell us what you did and what you found to be the result.  Then you can intelligently express an opinion. 

This author suggests that such views of digital are a feature of ingorance of sampling theory and that 192k has the potential for harming the sound.  I wish someone more up this this would comment on this very divergent viewpoinog. 

It's also simple to test his theories for free if you have a 24/192 or 24/96-capable DAC.  Just download some free HD music and listen.  Barry Diament and his Soundkeeper Recordings offers free sample tracks, "from the same album, same mastering session, etc."  for this exact purpose here. 

You can also read about the importance of higher bit/sample rates and digital filters here. 

In my mind there is one key aspect of any comparison between DSD and PCM that is not yet clear (to me, at any rate). 

DSD is a subset of the generic format Sigma-Delta Modulation (SDM), just as Red Book CD is a subset of the generic format PCM.  In more than 99% of all PCM DACs the actual digital-to-analog conversion is done in SDM format.  It is simply easier to do it that way from an engineering and cost perspective.  The incoming PCM is converted to SDM using a "filter" - but this is not what you or I would understand as a filter.  It is a very complicated conversion algorithm. 

Likewise - as I understand it - in more than 99% (and it could actually be 100% - I would like someone to clarify this) of all PCM ADCs, the actual analog-to-digital conversion is done in a SDM engine, and the resultant data stream is converted to PCM using an analogous conversion algorithm to the one mentioned above. 

It is important to note that SDM and PCM store the music data using completely different representations, and there is no mathematical equivalence between the two.  Therefore conversion between the two formats is inherently lossy.  Powerful algorithms can reduce the losses for sure, but from a mathematical perspective my understanding is that the lossy aspect is fundamental. 

So the issue for me is that in any real-world qualitative comparison between DSD and PCM, is it possible that all we are hearing is the effect of the SDM/PCM conversion algorithms?  In which case PCM will always give something up to DSD. 

So the issue for me is that in any real-world qualitative comparison between DSD and PCM, is it possible that all we are hearing is the effect of the SDM/PCM conversion algorithms?  In which case PCM will always give something up to DSD. 

Hmm.  My immediate response is - NOS DACs would be the 1% since they avoid the PCM > SDM step, so in this case you could then say that all true NOS DACs (that avoid PCM > SDM) should be closer to DSD playback...While I do not have extensive experience with NOS DACs, I'm not so sure that this would be the case as their presentations vary widely from DAC to DAC. 

That said, I very much enjoyed your thought-provoking post and will have to think more about it! 

I'll summarize my understand of it like this: Delta-Sigma is easier to perform analogous conversions and like you say, that's why you find it used.  But PCM has always been easier to edit and distribute. 

Not to dispitue your findings, but they are perfectly matched to what an a priori assessment would have concluded.  However, in my readings of various audio publications, I have several times run across articles outlining the noise generated by the DSD process - a noise level that increases in direct proportion to frequency.  In fact, I think John Atkinson wrote about this phenomenon a year or so ago claiming that DSD will forever be inferior to 96k PCM.  See <http://www.craigmandigital.com/education/PCM_vs_DSD.aspx> for a presentation of my dilemma.  Please be very aware that my mind in not made up on this subject; and in fact the degrees of separation may well be far less than the "damage" done in the recording, mixing, mastering process. 

What Monty's essay and its references don't account for is conversions. 

The conversion between DSD, which is in the native one-bit "language" of virtually all chips used in A/D and D/A converters these days, and PCM was referenced in a couple of the comments here. 

Another conversion that has occurred for PCM input in the chips used in nearly all DACs for the past two decades is "8x oversampling."  In order to avoid audible aliasing errors from the filtering necessary to convert digits to music, the conversion from digital to analog is done not at the input sampling rate (44.1, 96, etc.),  but at "8x" rates - 352.8 and 384kHz.  8x rates are attained in three rounds of 2x conversion, if necessary.  44.1 and 48kHz inputs must go through all three rounds.  But 88.2 and 96kHz inputs only require 2 rounds, and 176.4/192kHz just one. 

Each conversion, even though it is just a simple 2x "multiplication," requires a mathematical filter.  The way these filters are accomplished is by means of something called Fourier transforms.  A property of Fourier transforms is that the more you optimize the performance of a filter in the time domain, the less optimized it is in the frequency domain.  So no filter can be perfect; every one (and therefore every conversion) involves tradeoffs.  Thus your DAC performs this filtering, and time/frequency domain performance tradeoff, three times for a 44.1/48kHz input, but only once for a 176.4/192kHz input. ( And yes, this is true as well in most DACs that say they don't "upsample."  There are relatively rare examples of DACs, R2R "ladder" DACs like those from 47Labs in Japan, or the Phasure NOS DAC, a one-off design, that don't use 8x oversampling.) 

So that's the actual science, the objective fact, of what's going on inside your PCM DAC.  The question then becomes whether these performance tradeoffs are audible in two situations: Michael's, where the source material has been recorded at different sample rates; and earwaxxer's (and mine), where we've used sample rate converters in the computer (Sox in earwaxxer's case, iZotope in mine).  As reported by both Michael and earwaxxer, and in my own listening experience, the answer appears to be yes. 

Certainly in Michael's case this should not be surprising, but what about earwaxxer's and mine?  Actually, it shouldn't be too surprising there either.  The computing power in today's PCs, combined with the ability to optimize filters as easily as the next software release, means the filtering done in software in a computer can very possibly be better (less audible effect) than that done in firmware in a DAC chip. 

Thanks for that very well considered comment, judmarc.  Your point re: 8Xoversampling occurring in 2x steps (where necessary) reminds me of something Charlie Hansen talked about in our Q&A:

The advantage of performing the computation in a single pass is that there are always rounding errors at each oversampling operation.  If you perform it all at once, the rounding errors are minimized.  But if you perform 16xoversampling as a series of four 2x stages (the normal way, as it is the cheapest way), then the rounding errors are compounded four times. 

I've done DSD, vs other hi-res and Redbook testing at home.  Not truly scientific, but blind tests where I randomize a playlist and then try to guess what version I'm listening to. 

Results: the biggest difference for me vs. Redbook is DSD.  It just sounds a little warmer and more natural, not forced, and with more detail and space.  I can accurately pick it out more than 50% of the time, so for me no other testing is necessary. 

I've also put on some of these mixed playlists when listening more casually, and even when in the next room (open floor plan) and not in a direct line to the speakers, I can pick out the DSD.  It just sounds different (better to my ears). 

I have no problem in principle with someone doing ABX testing, but I'm not sure the results will mean anything.  My experience is that we are talking about small differences between the sounds of the formats.  Noticeable, but small.  Even "audiophiles" who don't have much experience may not pick out the differences.  Like most listening, it is a learned skill.  With experience, you learn to hear the differences. 

Once you've trained your brain to pick out the differences, you hear them when you listen.  This isn't some kind of illusion or bias, just training your brain to notice certain sounds (or lack thereof).  I've had the same experience with friends who say mp3 sounds the same as a CD.  It does - to them - until... I point out the differences on a few songs, and get them to notice.  Once they've trained themselves to notice, they can pick out the difference between mp3 and CD even on material they are hearing for the first time. 

DSD has a quality that I do hear from PCM which I tried to describe in this piece.  I also had an opportunity to hear a number of DSD sources in different systems at CES and they only served to reinforce this opinion.  There's an organic quality to the sound of DSD playback that I do not hear from PCM that's more akin to analog tape.  This holds for native DSD recordings, which are admittedly few and far between, but also for analog tape to DSD recordings. 

That said, I'd like to stress that the quality of the original recording is the most important factor, and ones attachment to the music obviously even more so.  So I look at DSD as simply another tool in the computer audio playback tool chest.  I've heard great sounding CD rips, wonderful 24-bit 44.1/48/88.2/176.4/96/192 downloads, and superb-sounding LPs.  As I said in my recent AWSI for Stereophile, computer audio and a turntable gets you the best of both worlds. 

Michael, thanks for this article.  I recently acquired a Matrix X-Sabre DAC (ESS 9018) that is capable of both DSD64/128 and DXD (352.8KHz).  I began searching for some samples and came across this site: http://www.2l.no/hires/index.html All of their files are from the same master DXD so it proved a great resource, and for free!  I picked up a few of the tracks in both DSD64 and DXD to compare.  I was really impressed by both, but quickly I found differences in the way the sound was presented.  Where the DSD64 track featured violin sections that sounded slightly compressed as if it was a single player but slightly dithered, the DXD track revealed separate players coming together as a whole.  Perhaps if the DSD were 128 it would even the playing field, but I'm wondering if this is more of a difference in format.  This could of course also be how my DAC is processing the data. 

Congratulations on your new DAC!  One thing to keep in mind with the 2L samples is they all are sourced from a PCM format, DXD.  I'd suggest trying one of the free samples from Channel Classics which are DSD through and through. 

I recently compared DXD to DSD but not with the same source material and I talk about it in an upcoming review that will be published very soon.  The bottom line for me is each format can prove to be musically engaging whether we're talking about CD-quality, DXD, or DSD so having a DAC capable of handling all of these formats is the real win win. 

Thank you very much for your great informative article.  I have a question.  It seems that you compare PCM and DSD played from the same DAC.  But what about PCM playback from really expensive DACs?  Is the difference of sound between 192 PCM played from an ultra high-end DAC, to the DSD played from a mid priced DAC that great?  I am asking because I recently acquired a Weiss MEDEA+ DAC.  The quality of playing high rez PCM from this DAC is stunning and I really cant think of what more can be done for digital sound to sound better than that, so real and analogue like.... But it does not play DSD and personally haven't heard DSD played from a native DSD DAC.  So is it worth buying a cheaper DAC to play DSD files natively on it, or my DAC plays PCM so fine that there is not noticeable difference.. Can you please comment on that? 

As a computer engineer, HiFi audio is a big hobbie of mine, I am trying to figure out the possible causes of the perceived difference between 94/24 and 192/24 in PCM. 

Let me start by saying that I do believe in the value of high resolution music, and I can hear the difference between 44.1/16 and 96/24. 

This makes sense to me.  The difference in sample word size between 16 bits and 24 bits is very large (2^16 = 65.5K vs 2^24 = 16.7M).  It gives you a much higher resolution to describe what is recorded.  To my knowledge, this makes the primary difference. 

The change between 44.1 KHz and 96 KHz sampling rate also makes a difference, but the difference is in the maximum frequency that can be represented.  The maximum frequency that you can represent/record on 44.1 KHz sampling is 22 KHz, on 96 KHz sampling rate it is 48 KHz.  While I know of no instruments that produce sound above 20 KHz, I know that there are some interaction harmonics produced that can exist above 20 KHz, and we are recording those (Note that I am intentionally staying away of the argument of whether humans can hear it or not). 

I read somewhere that is that even analog recordings, the tape bias is in the 30+ KHz range as its upper limit.  Which means that the maximum frequency that is being "recorded" is less than 40 KHz. 

The one question I have is in the perceived difference between 96 KHz and 192 KHz.  It might exist, as you found.  But it doesn't make sense to me. 

Most microphones can't record much above 30 KHz.  The high end microphones might record up to 50 KHz (ex.  Sennheiser MKH 800-P48).  Your DeVore Fidelity The Nines speakers specification say that their requency response upper range is 40 KHz.  They both should be well served with a 96 KHz sampling rate, and its maximum representable frequency of 48 KHz. 

For 192 KHz sampling rate to make a difference, there would need to be important information between 48 KHz and 96 KHz (max representable frequency by 192KHz sampling rate).  Even without getting into the argument of whether it can be heard by a human, I don't see how there is relevant information in the recordings above 48 KHz (microphone imited), or that it can be reproduced by the speakers we use. 

I know it is perception, and I am not trying to prove it wrong.  I am just trying to understand where the difference could come from between 96/24 and 192/24.  I realize that we might not have an answer, but in your opinion, what do you believe to be the possible cause of difference between 96/24 and 192/24? 

At least one member of each instrument family (strings, woodwinds, brass and percussion) produces energy to 40 kHz or above, and the spectra of some instruments reach this work's measurement limit of 102.4 kHz. 

So from one perspective, if we want to capture and reproduce a live acoustic event, for example, we would need to capture frequencies above 20k.  But frequency response is not the sole reason for higher sample rates.  In brief, filtering and noise shaping are better served by higher frequencies by moving these processes out of the audible range. 

Enjoyable read and what I would probably think too, but why only take off your glasses after you knew the difference between the sound of each file?  If a test subject feels the DSD128 'should sound better' than the DSD64, listens to each one, hears a difference and is later able to distinguish between the two, an original preconception cannot be excluded from tainting the results.  I understand this was not designed to find how to distribute food across the planet but it may have have saved you a bit of hate from some seemingly very angry personalities! 

NB Angry people; it's ok, Mr Lavorgna is just expressing an opinion, not testing the safety of therapies for your high blood pressure.  Isn't this meant to be for entertainment?  Get angry about poverty or climate change or something else that poses a risk to you and others, not how the author listens to his tunes. 

Hi, I follow a website called 'Trust me I'm a scientist.com' who has issued a golden ear challenge,

"Our promise to write a glowing, feature-length article about whoever becomes the first person on record to show he or she can reliably hear an improvement offered by any super-high-resolution file format under properly controlled conditions...It could very well be that 16/44.1 PCM or 320kbps MP3 represent the very pinnacle of audio quality as far as the human ear is concerned. ( At least that’s what the overwhelming weight of science seems to suggest so far.) 

But hey, it’s worth trying, right?  Especially since so many marketers claim there’s a real difference.  And if that difference is real, it should be acknowledged." 

I said I would let you guys at audiostream.com know about it as it might offer worthwhile coverage for audiostream were someone to take it up. 

I've read the article and all comments carefully and I have some maybe interesting insigths, some of them are based on my knowledge about data converters (especially sigma-delta type) and some are based on my experience (I'm only 30 then it's not large:)). 

First of all - listening experiments...in my opinion, every time we're evaluating sound quality with some "scientifically well-stated" tests (double blind, ab, abx, peaq etc.)  then we're getting results that are more related to conditions under wich we've performed given test than actually the sound quality...i think that results gathered from all of these tests are only some sort of approximation of how do we "really" hear the differences between sounds and sound features.  I wrote "really" cause we still don't know how do we really hear sounds (really without quotes :)).  What I want to say is that we should consider results from all of the listening tests with reserve and distance. 

Second - higher sample rates...apart from already discussed frequencies that instruments can produce, mics characteristics and what frequency range we can hear there is another reason which entitles usage of higher sample rates - digital filtering - http://www.cirlinca.com/include/aes97ny.pdf

Third - ADC and DAC converters.  There is an opinion that DSD sounds better than PCM.  But I'm a little bit confused...We consider here only the file format or the DAC architecture enclosed in sound system which have reproduced sound being subjectively evaluated? 

I think that there is some sort of misconception regarding ADCs and DACs nowadays.  Every manufacturer writes specification of his product and is proud of, for example, a bit resolution which is 24 bits actually available.  But it's not the real bit resolution of the converter - it's an outcome of standard equation connecting bit resolution and dynamic range.  True resolution is one or few bits in sigma-delta converters which are used commonly these days.  Of course they work with 64x or 128x oversampling (wow - it must be perfect quality :)) instead of 44.1-176.4 kHz in conventional converters (for example R-2R PCM DACs).  But the main difference between conventional PCM and sigma-delta converters lays somewhere else.  Conventional PCM converters have real 16-20 bit of resolution and they code bits or decode them all in one moment of time (creating digital words from bits - adc or decode them into electrical signal - dac).  In sigma-delta converter structure there always exist: interpolation/decimation filter (DAC or ADC converter), coarse quantizer (one or few bits of resolution) and of course a feedback loop which spectrally shapes the resulting quantization error (spectrally shaping means shifting spectrum of quantization error into ultrasound frequency range - it is called noise shaping which together with oversampling is the main cause of quite large dynamic range values of sigma-delta type converters).  Then this spectrally shaped error is substracted from the input signal and whole process repeats and repeats. 

Thus, sigma-delta converter output sound sample depends on actual input sample and whole history of processing, it's highly nonlinear (because of quantizer) and nonstationary (because of feedback) process (by the way SABRE DAC wich was mentioned above is sigma-delta type too, but with some algorithms which are trying to make converter less chaotic).  Recently I've perpetrated a PhD thesis about short-time analysis of performance of sigma-delta converters.  And it turned out that the error which is introduced by sigma-delta converter into output sound is always correlated with features of actually processed sound (it's amplitude, speed of change and shape).  These errors are larger when the sound has more transients, is louder and it's shape is changing faster.  Since I believe that we hear not in frequency domain but in time domain then this correlation is perceived by our ears in some way.  Of course conventional PCM converters are nonlinear systems too (there is quantizer, but with much more bit resolution), but their time-domain response is immediate (with respect to generally speaking design and element tolerance).  Thus, errors that they produce are much less correlated with sound signal which they're processing. 

In conclusion, I'm just trying to say that in my opinion the PCM system will always sound better, because modulation errors, that always exist, are much much less audible in conventional converters than in sigma-delta converters.  Of course when making comparison of PCM vs. sigma-delta the music from CD must be played only by CD player that is equipped with real PCM conventional converter - in other way this type of test is pointless. 

So I go back to my question, were these comparisons that You performed (DSD better than PCM) concern only format or whole sound system? 

Hi Marcin, your concept of time perception vs frequency caught my eye.  I would say this: for all our evolution since coming down from the trees we have used our extraordinary time correlation abilities to pinpoint 3D spacial position by comparing phase differences in high frequency signals, with amazing accuracy.  In general the higher the frequency the more accurate the position fix, which allows you to know where a mosquito (or the snap of a branch under a tiger's paw) is to a great degree of precision.  This is a clear evolutionary advantage.  The musicality of a recording owes more imo to this phase coherence than to possible distortions which are more easily accommodated by the brain. 
So although PCM might in mathematical theory be as good or better in terms of amplitude distortion, that is less important for the appreciation of fine sound than the phase issue. 
Musicians in my studio cannot generally resolve time differences of between 20 and 2 mS on their own (without a simultaneous reference) but once we get shorter than 1 ms (remember the difference between your ears is about 0.6 mS) then the phase correlation kicks in and is best perceived between 0.5 - 0.2 mS or 2K - 5K in my general experience. 
Whereas tape audio phase accuracy is determined by the head alignment and is relatively linear for a given machine, as is vinyl reproduction, in PCM D-A's the subharmonic phase excursions clearly seen in a sweep analysis and predicted by elementary hi-Q filter theory create a highly unstable stereo imagery that sounds 'flat' 'lifeless' like hitting a wall' and so on, because each part of any percussive sound and anything that has natural harmonics extending up to an over 20K is smeared across the audio picture, to say nothing of the subtractive subharmonics in the audio spectrum from sounds in the 20K - 200K range that we don't hear directly.  All out good old tape machines were more or less flat up to 50K with 150K bias, as were the mics and pre-amps, and they sounded great.  The only time we reduced the HF response was when building a studio near a radio or TV station transmitter where things might get ugly. 
Hope this is of some help and if you want to talk more email me harry@springstudio.com.au...

Also really like Firedog's comment about half way down here, that the differences are small, and need practice to hear.  Interesting that he's also using a Mytek, so looking into those for my personal test. 

Ive been recording fine music for 35 years and a few years ago I bought the Korg mutiformat portable recorder M1000.  Since then I have lain in the mud recording galloping horses at 2 metres at 4 am, birds, waves, jets, choirs, orchestras and all the rest on everything from 44.1/24 to DSD 5MHz and can honestly say there is no comparison between PCM and DSD.  Having cut 5000+ LPs on acetates and QC's the results, Ive had a wide experience of recordings, and if I had to compare DSD to anything it would be 30ips 1/2" or 16 track 2".  The phase coherence is what makes the spacial information 'work'.  The filters necessary to remove the sampling frequency from PCM create subharmonic (and audible) phase swings in regular patterns even at 192K, though that is much better than 44.1.  In desperation I am about to release selected tracks from this lifetime of recording as PONO-compatible tracks but my first choice for physical archiving is DSD.  I am not alone in this ....

If the up sampling of PCM is in multiples of 44.1kHz, then why isn't 176.4/24 the best hi resolution file to distribute and play.  The 96 and 192kHz must need some complex algorithms to fit into the DAC frequency of 8X44.1? 
Many DACS convert DSD to PCM to decode and something must be lost in there, although, according to a previous post all DACs convert PCM to single bit for decoding, so does it imply multiple conversions?.  I suspect we need to know exactly what out DAC's are doing and what file types they handle naturally and then get music in that format.  I have been told that DSD should never be converted to 96 or 192, because of the sampling frequency, it's natural format is 176.4kHz (multiples of 44.1kHz)? 
On a similar vein, many think that lp's cut from digital recordings are no better than CD, I have been thinking about this because the lp's often sound heaps better, I suspect this is because the lp may have been cut from the hi-res digital file, which is then "castrated" to make the CD. 
If you know of an article that describes exactly how these formats relate to different DACS, I would be most grateful for the link. 
I don't need convincing lp is better than CD, and Hi-res digital is better than CD period.  Currently recording some rare lp's in DSD... very impressed with the results (KORG MR2000s), just not sure what format to standardise on.  While the Korg still works I'm using DSD(5.6), but what if I get another DAC, I'd rather record in the format I'm going to use than convert it from one format to another on the computer....or am I worrying about nothing? 

Here's a PCM Sine wave of 22050hz Sampled at 44.1khz. 
note that anything that's not the actual sample, is the DAC 'guessing' trying to connect the dots
( yes.. the DAC will be connecting the Dots differently than this drawing, and that's beside the point )

The wave is described by a single sample per half-wave, this is the minimum amount of samples per half-wave required, resulting in a brick wall at 22050hz. "  but isn't this outside the audible spectrum? "  Well let's do a 11khz example then. 

2 samples per half-wave at 11000hz.. No amount of bits can change this, even at 10000 bits /44.1khz the waves would still consist of the exact same amount of samples. 
Then we have 4 samples per half-wave at 5500hz, 8 samples per half-wave at 2750hz and so on. 
When describing a sine-wave using 2 samples, increasing the bit-depth will only improve the 'accuracy' of those 2 samples, but i wouldn't refer to it as a higher resolution, since there were no increase in samples. 
Higher Bit-depth allows the 2 samples to have a more accurate amplitude, but 2 samples will still only be 2 samples. 

Bit-depth = SNR
Sample rate = resolution

I've heard people refer to the 'kbps' of a recording, as if that is the "Resolution"
you calculate the kbps by saying 16bit x 44.1 sample rate = 705.6 x 2 = 1411.2 kbps (regular CD)

( multiplying the result by 2, because it's stereo )

So what's being done is multiplying the SNR with the sample rate, and they believe the kbps will represent the 'resolution'
That is like claiming a monitor's resolution were 120hz refresh rate x 1920x1080pixels = What? 
SNR and Sample rate are two different things, multiplying them is a pointless exercise. 
You might have a computer screen with a refresh rate of 800hz, but if it's maximal resolution is 640x480, that's still a low amount of pixels, even though it's a very fast refresh rate, multiplying the refresh rate with the amount of pixels, will be the exact same pointless exercise. 
DSD's entire worth of kbps is in samples, there ain't no Multiplying SNR with Sample rate when it comes to DSD.  So comparing DSD and PCM's amount of information "kbps" is ridiculous. 

What did we learn from this?  that 16/96 sounds better due to having more information?  I'm afraid that ain't the case. 

32/44.1 SNR of 192,5dB and a sample rate of 44.1khz. 
24/48 SNR of 144.49 dB and a sample rate of 48khz. 
16/96 SNR of 94 dB and a sample rate of 96khz. 

The SNR of expensive Dac's is around 120-130 dB SNR and amplifiers are well below that, so there is no point of 24bit or 32bit, roughly 20bits would be sufficient, due to the limitations of SNR in dac's and amplifiers, cables etc. 
Which is also the reason for why they are sticking with 24 bit-depth and are only increasing sample rate. 
There is simply no point in 32bit stereo audio. 

Another thing is that when you decide on having a static sample rate, it's an inescapable fact that lower frequencies will consist of a larger amount of samples than higher frequencies will. 
When you choose sample rates down in the 44.1 - 192khz your 10-20khz audio spectrum will digitally speaking be a square signal trying to describe curved waves. 
The PCM-Dac will be playing 'connect the few dots' while the DSD-Dac are producing so many dots, that the dots themselves will be drawing smooth curved lines, resulting in a pure analogue signal in the digital domain. 
How people fail to comprehend this is beyond me..

I've heard people say, and it seems to be a regular saying among PCM fanboys, that 24bit / 88khz is 'the same' as 2.8mhz DSD... how they end up at that conclusion based on kbps is laughable. 
88khz sample rate will describe a 20khz sine wave using 4.4 samples, how are these 4.4 samples as good as DSD64's 2.8mhz 140 samples at 20khz?  Oh yeah that's due to the fact that they multiply the SNR with the Sample rate, because they are PCM fanboys who needs to go study the subject more thoroughly. 
There exists actual purchasable music at DSD 11.2mhz sample rate, at such high sample rates even a 20khz sine wave will consist of 560 samples. 
16/44.1 needs to be all the way down at 78hz before it starts having that many samples to play with. 

Then let's do a PCM 24/96 example
24bit x 96khz => 2304 x 2 = 4608kbps
Please understand that we are only dealing with 96kbps worth of samples.  Multiplying the 24bit SNR with the 96kbps sample rate gave us 4608kbps.  Pulse code modulation is a very inferior way of handling audio, it's a code where the bit-depth tells the amplitude of the samples, and the sample rate provides the samples. 
Producing multi-bit PCM chips capable of worthwhile sampling rates has not yet been done, during the roughly 36 years PCM chips have been the standard, which is why i refer to PCM as being inferior. 

In reality DSD have a very low SNR since it's only 1bit, which in theory would only give a 6dB SNR. 
By using noise-shaping they increase the SNR way up to 24bit standards, without having to imbed that method into the digital signal, it's strictly hardware based, which is very advantageous. 

11.2mhz quad DSD example. 
1bit x 11200 => 11200 * 2 = 22400 kbps

24 bit / 192khz sample rate example
24 x 192 => 4608 * 2 = 9216 kbps

So if we were to assume ( which we shouldn't ) that kbps said it all, then Quad DSD has 60% more 'information' than 24/192 pcm, and should sound 60% better, but that's not everything..
11.2mhz DSD has 22400 kbps worth of infomation in pure sample amount!  while 24/192 only have 192kbps worth of samples, that's less than 1% of Quad DSD's sample amount measured in kbps. 
The fact is that DSD sounds so good that i bought a 1000 dollar DSD DAC immediately after hearing DSD for the first time, with no hesitation at all. 
There is no comparison, even DXD which is 24 / 352.8 PCM sounds harsh compared to 2.8mhz DSD. 
(Yes my DAC also plays DXD) Currently using a TEAC UD-503 Quad DSD / DXD capable DAC. 

The studio's that owns DXD ADC's who actually record music at 24/352.8 says themselves that their recordings sounds best at DSD even though it was recorded in DXD 24/352.8.  And i agree. 

"Interestingly, we found after listening to DSD files made from DXD edited masters that the DSD, particularly the higher bit rate examples, sounded more natural, spacious, and life-like than their DXD parent from which they were made." 

Yes, obviously 11.2mhz sample rate sounds better than 352khz sample rate ..?  i would'n need to hear those recordings to arrive at that conclusion.  You can also buy actual music recorded in DSD on that site, either you buy Analogue to DSD, or you buy DXD to DSD, but you stay far away from PCM, once you get the chance! 

People who writes down obscenities like " DSD and PCM sounds alike "
These are bitter people who have never even heard DSD... i can guarantee you that, there is no resemblance between DSD and PCM!  I've read somewhere that blind tests had been performed and nobody were able to differentiate between DSD and PCM, and that's just impossible, where did they find these people!?  That would be as unlikely as finding a target group incapable of differentiating between tube amps and class-D amps...

All PCM sound alike, since the difference between 44.1khz and 192khz sampling rate is too small for it to be audible. 
The difference is only 147khz.  The difference between 2.8mhz and 11.2mhz is 8400khz, that's not a small difference, and it's very audible.  In the world of PCM the different sampling rates you get to choose from are just too closely related for there to be an audible difference, i don't think i would be able to tell them apart in a blind test.  Whether the 20khz sine wave is made up of 1 or 2 samples, it sounds harsh in both cases, this is why the difference between 44.1 and 96 is inaudible. 
It's a whole other story when it's between Sine-waves consisting of 280 samples or 560 samples. 
That won't be 2 similar cases of harshness. 
Well i'll leave it at that. 

The biggest difference from a production viewpoint is the low level coherence of reverb and decay tails, which if 16 bit feel artificial when they disappear into the dither.  Huge difference between 16 and 24 there.  We regard the bit depth improvement the most important advance in recent years. 
While 32 bit might seem mathematically useless to you, when its done as a floating point calculation, there is a lot to be gained from running a DAW in 32 or even 48 bit mode, internally.  Almost everyone does this. 

Trying to compare finished recordings is a bad way to evaluate audio experience, unless possibly the source could be an ideal live performance, maybe an orchestra in a great auditorium.  Then splitting the signal into several complete recording systems might yeaid some really interesting results. 
In fact I think comparisons are difficult at the best of times, except when its completely obvious.  As a producer for 40 years, every so often I meet a new piece of gear that does something wonderful. 

My first Nagra recorder was like that - the pre-amps were simply magical and so very musical.  Suddenly low level layers of obscuring distortion were absent , a revelation.  Using my DSD recorder live alongside a conventional 48/24 recorder was the same - amazingly obvious improvement. 
If there was some affordable way to multitrack and edit and process DSD I'd be right there.  As Im sure you know, some systems revert to making in intermediate 384K copy of the edited section which seems crazy.  But how else to meet two diverging signals? 

Heres a hard question for you.. I cut lacquers on a Neumann VMS70.  How come all the clients think the vinyl sounds so much better than the PCM source files?  Id be interested to hear your thoughts on that one! 

Everyone I hope knows that to convert PCM (RedBook) bit perfect, it has to be done using an R2R ladder Multibit based dac.  Not converted on the same Delta Sigma based dac that was used to convert the dsd.