DSD mastering and quality

One likely last post in this thread from me, because I think we’ve worn this out. But, given that Pure DSD256 recordings are not economically viable and no one purchases them, I thought I’d just do a quick check in my music library database of how many Pure DSD256 or DSD128 recordings I have that have been recorded originally in DSD versus a transfer from analog tape. These include those mixed in DSD and those mixed in analog from the microphone preamps and then fed directly to the A/D converter for capture. I counted 106 (NativeDSD lists 212 in their catalog). Next week the number will likely have increased with new releases coming out. There must be some rationale for why the labels releasing these continue to do so. Here they are. (If I were to add all the Pure DSD256 transfers from reel-to-reel tape, the number would be vastly greater.)

(Click on image for larger size.)

Well as far as I understood, there can be DAC’s which do prefer the DSD format, even though it was just converted back from the mixing format DXD….could be possible…however, a suspicion remains that the conversion back to DSD is just done to imply a meaningful complete DSD path.

I think the DSD lobbyists would do well to be as transparent as possible with what makes sense or not…on the other hand, this transparency is hardly marketable…a difficult situation.

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Didn’t understand this. (To be honest, I don’t have the patience to read posts longer than two paragraphs, so haven’t read most posts by @rushton or @stevensegal which might have informed me in understanding this. Sorry.)

Are you saying that it doesn’t make sense for DACs if the content was mixed in DXD? Or other way around?

@shankha, two things: 1) generally buy the format closest to the edit master, 2) test file formats on your DAC because some DACs will play PCM better, some will play DSD better, and some DACs just don’t care, depending on the design of the DAC.

In my main system, when the edit master is DXD, I’ll download the DXD file because it will generally sound best. (Fewer conversions have already been made to the file before my DAC does it’s own thing processing that file, and fewer conversions mean fewer unavoidable conversion artifacts.) But on my wife’s office system, her DAC seems to prefer a DSD file and will generally sound better with the label’s already converted DSD256 format file from that same DXD edit master.

So, my mantra has been: listen, listen, listen to find what is best for you.

I’d be curious what an accomplished record producer and mixer especially in DSD like @cookie has to say about all the back and forth between @stevensegal and @Rushton. And especially since she releases in pure DSD and mixed in DXD.

Cookie, in case you have the patience, start with @rushton’s post beginning with

Yeah, this is the principle I follow as well. But I do tend to lean towards DSD256 rather than DXD masters when it comes to forking out money.

My DAC is the PSA DSD which likes DSD anyway. Plus it basically upsamples everything including PCM to DSD256 anyway first.

Yes, my DAC does the same (Playback Designs MPD-8). But, it does a better job converting to DSD256 than does Pyramix. The edit master files through my MPD-8 always sound better than the Pyramix DSD256 output files. The difference is subtle, but it is there and pretty consistently so. You might find the same with your PSA DAC.

Hello all, I haven’t had a chance to read all the posts about DSD vs DXD but I am aware of the arguments presented so…

Last November we decided to prepare workshops to answer this question. The first step was creating files for those interested to compare. So, we created various DSD256 and DXD files for you to listen to. Here’s a coupon code for reducing the price.

PaulsPSPeeps$10

Here’s the link to purchase the files.

It’s not cheap, but we plan on offering the Workshop in April and the cost will be offset by the price of the comparison workshop.

We’ve had several people listen and we do hear difference, albeit very small. I know what I prefer and the rest of our blindfold test crew… but I’ll you you decide what you prefer.

Doing the test was an eye opener as to how Merging Technologies decided to setup DXD and DSD. More on that later. I believe we offered enough comparisons to overcome their software playback decisions.

I’ll be back next week,
Cookie Marenco
Priducer and Founder
Blue Coast Records and Music

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Unless @Cookie contradicts, in my understanding she usually releases analog mixed DSD, maybe some unmixed, pure DSD releases, but she doesn’t prefer DXD mixed DSD releases and possibly just creates them for comparisons.

Yes, I mix DSD256 through an analog console back to DSD256. I don’t mix in DXD (unless I’m paid for a “not Blue Coast Records” project. :slight_smile:

Cookie Marenco
Blue Coast Records (my label) and Music (the store with many labels).

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These comments suggest that you might be over-indexing on the 1-bit-ness of DSD, but that’s not a criticism – it’s just another example of why I hold a general disdain for the activities of marketing departments in any technical context.

Repeating a comment I made in a different thread recently, the 1-bit sampling of the DSD format is like jitter in that it truly only matters in precisely two places: the A-to-D conversion of recording and the D-to-A conversion of playback. And it only matters in those places because it allows us to build more accurate devices from physical materials, ie you can get closer to perfect linear responses switching one device between just two states than you can when combining multiple devices to represent four or more states.

Once you’ve done the A-to-D conversion and you’re in the purely mathematical realm there’s absolutely nothing lost by representing the signal with other symbols of as many bits as you care to use. In truth, multi-bit representations provide a more intuitively correct representation of the signal than DSD itself does, because in DSD 1 and 0 don’t mean “on” and “off”, they mean “positive” and “negative” values of equal magnitude. Zero isn’t negative the last time I checked. But I can use any multi-bit signed numeric system, whether integers or floats, and transcribe the DSD waveform into equal-valued positive and negative samples and the signal remains utterly true to the original.

Multi-bit does not mean “chopped-up”. It means nothing more than the ability to define the amplitude of the waveform at a particular point in time with more precision than just “positive” or “negative”. It’s that extra precision which allows for mixing, by making the DSD waveform of track 1 smaller or larger relative to track 2 and then having enough room to add them together without losing any of their individual contribution.

If we want to deliver the final product as DSD (again, because that’s potentially beneficial for D-to-A conversion using physical devices) it does have to go through another sigma delta modulation process to resample our mixed high-precision waveform back to a 1-bit representation. But guess what…?

That’s exactly what happens when you mix DSD using analog systems too! And I promise you the pure digital approach has less noise and other distortions than the analog gear necessarily introduces.

The reason it hasn’t been done routinely in the past is not because it was impossible or sonically compromised but because it didn’t stack up in terms of cost/benefit. Most entry-level audio work is done with 96kHz sample rates. Working at DSD rates is 30, 60, or 120 times more demanding on your workstation if you go all the way to 4xDSD. So people make trade-offs and do the one thing which actually does damage your original DSD waveform: they downsample to lower sampling frequencies. DXD, which is a very high res PCM encoding, is only four times the sampling rate of 24/96.

But please note carefully: it’s the reduction in sample rate, not the use of more bits per sample, which does the damage to the original DSD signal from the recording desk. Single-bit-ness is only valuable right at the beginning and right at the end of the process.

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Thanks cookie!

Thanks for the just the facts explanation.

Question: what PCM encoding of 4xDSD would be considered NOT downsampling? It needs to be at 11.2Mhz? Is that what you’re saying?

No, because remember, 11.2mHz is for 1 bit.

Now, divide that number by 24 bits to get the answer, when using PCM.

It’s something like 466kHz 24 bits (which is very close to 352.8kHz 24 bits). Or, put another way, 352.8kHz times 24 bits is a DSD single bit speed of 8.4mHz.

This is one reason why a number of smart people I know are pushing us to switch over to 2X DSD and then mix in DXD.

I am still sticking with what sounds best right now.

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Hmmm, this does not sound like what @tedsmith was saying in this thread:

But, then, he was getting very theoretical in this thread, focused on the directions things needed to be going.

I like your philosophy: “still sticking with what sounds best.”

Great thank you @Paul

Looking forward to 768khz!

This may be one current advantage of analog mixing…although there are 2 DA-AD conversions instead of two digital “conversions”, resolution of analog is “infinite” :wink:

The most interesting part of this topic would be, why e.g. Cookie clearly prefers the opposite of what Paul prefers. As I assume both hear roughly the same, the question would be: what’s so different regarding their setups?

My assumption is, the digital parts (Pyramix) are more similar than the analog gear used in their experiments.

Sorry @Paul, but that’s not right – or at least it’s only a right answer to a different question.

Yes, I would agree with that. By definition, any reduction of sample rate is “downsampling”.

There are two dimensions to think about with digital audio: the sampling rate, and the precision of each sample which is often called the sampling depth and measured by the number of bits used. The main point I’ve been trying to convey in these comments is that PCM and DSD (and PWM/PDM) are not as completely different from each other as people tend to think – and you can do some remarkable things when you understand their commonality.

The #1 rule in digital audio is that your sampling rate must be more than twice the frequency of the highest frequency in your signal. If your signal has frequencies higher than half your sampling rate then your reconstructed audio will contain noisy reflections of those high frequencies folded back down into the signal band that you want to be pristine and perfect. For this reason, digital audio relies heavily on the use of low-pass filters, which as their name suggests permit low frequencies to pass through easily but become increasingly restrictive to signal components as their frequency rises.

LPFs can be implemented with physical circuits to filter analog signals during recording and playback, and they can be implemented with pure mathematics in the digital realm. Doing filters digitally can be dramatically more efficient (component cost, board space) and more accurate. But every filter comes at a sonic cost: they distort the phase relationships in the signal by causing a delay that’s not evenly applied. Higher frequencies are delayed more than lower frequencies. And the steeper the filter, the worse the effect.

Why am I going on about this?

Because it’s the heart of your question, and why Paul and Cookie are avoiding DXD. If you want to reduce the sample rate of some digital audio that you have (eg, some DSD) then first you must filter the existing signal so that no components are present greater than the mid-point of your target sample rate, and this impacts on phase relationships at high frequencies. Coming down from DSD256 to DXD is a big drop, reducing your ultrasonic headroom (by which I mean the sampling frequency space above our nominal audible band cut-off at 22.05kHz, the space in which a filter can operate) from 8 octaves to just 3. Now 3 octaves is still a fair bit of headroom by PCM standards but if you can hear degradation from DXD processing of DSD recordings, that filtering process and its necessary counterpart when converting from DXD to your final delivery format are the most likely explanation.

What Ted does in the DS DAC family, and what I presume the HQPlayer folks are doing, is processing the signal – even if just for volume adjustment – at the full DSD sample rate and therefore not having to use any low-pass filters beforehand. This leaves your phase relationships intact while you do mixing. Note that any EQ also impacts phase relationships but since we’re usually only making a few dB of adjustment the impact is much less than a filter which goes many times harder than that.

The challenge for everybody is to get their heads around how that is both “PCM” and “DSD” at the same time.

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Although we all know we’re splicing hairs and that recording quality rules out all those differences by far, it’s still very interesting to have the basic and theoretical understanding right.

Imagine, we’re here with the number one DSD experts as some of the most interested folks in this regard and anyway it takes countless pages of discussion to get to the ground.

In the real world, you have to find someone who just heard of SACD’s and may know that it’s based on DSD.

Then you’d have to start explaining a day or two, how contexts really are with one of the more simple and early conclusions that most of what this person listens to on SACD’s from most labels is probably just DSD converted 24/96 PCM recordings.

It’s similar but a bit more complicated than with buying the right kind of records :wink:

I would assume that as Cookie has been in the recording business for 40+ years and probably knows analog better than the back of her hand, that remains her preferred method of editing. Paul’s new to the recording game and finding his feet, but clearly wants it to be as DSD as possible, whatever that is.