Michael Lavorgna

This is depressing.

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.

In general, my experiences mirror Michael's.

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.

Any thoughts on how close a 24/44,1K file would be to a 24/96K?  In other words, does the increased bit depth have more of an impact rather than the frequency? 

It's intersting that some labels -- BIS is an example -- make such a big deal about "24 bit" recordings but don't focus on frequency much at all.

Suggestion for an AudioStream interview:  contact Richard von Bahr at BIS and ask why his lable is still doing most of it's hiz-rez recording at 24/44,1.

When I was doing some of the mastering for FIM, many of the files were "only" 24/44.1

Even though it was 44.1, the files were some of the best I've ever heard. I'd much rather have a 24/44.1 file to work with than 16/352.8kHz file!

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.

Since so many modern recordings are PCM, is it worth converting them to DSD if that becomes the dominant HD distribution format?  Or does the 24/192 file sound better unvconverted?

If you could convert your 24/192 file to DSD, and compare the resulting DSD file with the 24/192, it would be interesting to know what differences in sound quality exist.

Hi Michael,

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.

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.

Nice tests, nice post. Really enjoyed the read. 

There's absolutely nothing surprising here as going right up the resolution scale, the output improves, given all files created from the same source material.

What is interesting is your perceptions of HOW they differ. What get's "added back" are exactly the dimensions I am constantly trying to get back in my mostly-redbook playback chain. 

So - perfect correlation there!

Aside from all this, and trumping all, is the original recording quality, as we all acknowledge. I have 24/192s here that sound no better than 44/16 and it's due to the source.

I've got some unbelievable sounding 44/16, as we all do, and some 44/16 that's so bad it's unreal.

One big uniform solution is probably not in the pipeline during our lifetimes. So we will continue to scrap it together best we can for now and enjoy the good recordings we can find!

http://music.e-onkyo.com/

Use Google to translate.

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).

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.

http://people.xiph.org/~xiphmont/demo/neil-young.html

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.

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.

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.

Hey Michael, your post is highly informative. But i´ve read it 3 times, and sill coudn´t understand if you think that DSD64 is better sounding then 24/192 PCM.

You said that any HD PCM and any DSD is better than CD. Witch I agree positively with.

You said that 24/192 is a little better than 24/96, which I felt the same using my EMU 1616m DAC at home with ADAM A5x monitors.

You also said that DSD128 is a little better than DSD64, which is problably true, due to more information captured by DSD128.

But Do you think DSD64 is better than 24/192 PCM, or they are similar to each other?

Or 24/192 PCM is similar to the sound quality of DSD128?

Of couse, I´m refering to your findings in this specific listening test.

16/44.1  << 24/96 << 24/192 ????  DSD64 << DSD128

Thank you

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.

Have you compared DSD to DXD? If so can you comment?

Michael,

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?

Thanks again for the time you took to write such a great review!

Michael,

Thank you for the article. I found it very intersting.

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?

Thanks, Juan

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.

Regards

http://www.trustmeimascientist.com/2013/10/07/update-on-the-golden-ear-c...

Hi and thanks for the article.

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?

Thanks in advance for the answer.

Marcin

Enjoyed reading the article, and found a lot of interesting, helpful, and a bunch of not so helpful stuff in the comments.

Soundkeeper files very useful.

http://www.soundkeeperrecordings.com/format.htm

Really interested by the dual 5.1, 2.0 various PCM and DSD on a DVD-A, SACD twin disk.  The music also sounds interesting too Liszt Grand Etudes however can't seem to get one anymore.  Anyone have tips for where to find, or an equivilant type download? (Soundkeeper just PCM) http://www.velutluna.it/index.php?c=7a&l=eng&spc=&id_cat=1&pagina=&id=101

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.

http://www.audiostream.com/content/dsd-v-pcm-file-comparison-16441-2496-...

Anyway, thanks for writing up, enjoyed it.

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 )

http://postimg.org/image/v1ty48z4p/

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.

http://postimg.org/image/m2v5j3wex/

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.

Here are three examples: 32bit/44.1khz= 2822kbps
24bit / 48khz = 2304kbps
16bit / 96khz = 3072kbps

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.

let's make a DSD SACD 2.8mhz example
1bit x 2800khz => 2800 * 2 = 5600 kbps
What we have here is 5600 kbps worth of samples, in other words 5600 kbps worth of actual resolution.

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.

https://referencerecordings.nativedsd.com/albums/FR718SACD-beethoven-sym...

"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.

-Hitashi

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.

Cheers George

Cheers George