DSD comparison to PCM.

You'll have to do some research if you want to understand this. 64fs DSD has a sampling rate of 2.8MHz. DXD, the Pyramix PCM editing format has a sampling rate of 352KHz. That's a 8X reduction, and requires a filter to eliminate all energy above 176.4KHz so as not to be reflected back into the audio band due to Nyquist Therm. RBCD sampling rate is 44.1KHz. That's a 64X reduction, and requires an even more drastic filter to eliminate all energy above 22KHz.

Decimation isn't an argument, it's the way this stuff works.

I'm in Boston also. Give me a call if you'd like to talk about it. I'm in the book. Tom Caulfield, Marshfield.

Thanks this is clearer, I am ware of most of this (didn't realize DXD is a PCM signal), but I'll give you a call anyway. My original question (a little rephrased) was: manufacturing costs aside, if you were able to record and playback in 176.4/24 PCM would DSD in any incarnation be solving anything better. Ragnar seems to claim all *hi-rez* digital will be the same. You seem to indicate that the Nyquist filters, at least those applicable in my argument, would still be detrimental (let's leave RBCD filters aside, we all know they are very steep and very detrimental)?
 
Bruno (from when he was at Philips and research in their labs on PCM/DSD) reported that to be able to take DSD and transcode to PCM transparently requires 32bit/384kHz.
Could be relating to both/either transfer function-truncation and noise shaping-filtering I guess, but one would need to ask him directly as I am not sure he has expressed why on any forum.

So technically for true transparency it would seem 32-bit/384kHz PCM is required, but that is relying upon whatever work Philips did in their labs and the scope of its context.
At this level it becomes more efficient to store-archive as DSD when considering datastreaming and storage.
Cheers
Orb
 
Oops was late should say 352.8kHz, hence the development of DXD.
Being vague as usual transfer funtion is complete aspect of any (re)quantization-lengthening-truncation.

Cheers
Orb
 
For storage, PCM is more efficient than DSD.

If you're talking about file size, it depends on the sample rate. You're correct if the equivalent DSD file is 2.8MHz, and the PCM file is 44.1/16 or 24. But a 192/24 PCM, not compressed with FLAC or the like is more than 30% larger. Bruce could give you exact figures for each sampling rate.

But efficiency is why telephone transmission is DSD like. Only the change information is transmitted each sample time, not the entire "stand alone" value of the level each sample time.
 
For storage, PCM is more efficient than DSD.

DSD64fs is the equivilent file size of 24/88.2, so I can not agree with your statement. I worked in DXD when I was doing FIM mastering and just 1 track would routinely be over 1GB! An album would be 12-15GB whereas same SACD would be 3-4GB.
Now when you start working at 32/352.8, then you will need RAID arrays and 100's of GB!!
 
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Thanks Bruce,
it is counterintuitive to most peoples ideas but at higher rate PCM, DSD definitely has its pros as a streaming/archive format over PCM.
Comes down to 1-bit vs 24-bit combined with sampling rate of PCM.
And then considering true transparency of 32/352.8 for PCM (when dealing with DSD) it really hurts as you mention :)

Cheers
Orb
 
What puzzles me are the very high bandwidths we're seeing here. Studio microphones have almost no output above 50 kHz; many are gone by 30 kHz. Above these frequencies, the output is both ragged and falling, with very uneven group delay since the diaphragms are no longer moving as a unit and are going into multiple modes.

Since professional mikes have so little output above 50 kHz, what acoustic signal content is illuminating the Nyquist-filtering group-delay errors that are part of 176.4 and 192 kHz PCM? Sure, an arbitrary square-wave test signal exposes the group-delay problems, but that is an all-electronic signal with no existence in the acoustic world.

B&K and Aco Pacific 1/4" instrumentation microphones have extended response (80 kHz) and well-behaved rolloff regions, but these are not generally used for music recordings.

In this example, we're looking at the edge-of-passband characteristics of the recording microphones that are commonly used in studios. If everything above 50 kHz is very low level, and what little remains is a collection of narrowband peaks with very rough group-delay variations, what effect does that have on the transmission system? That would seem to be an argument for Nth-order Gaussian lowpass filtering (which is more or less what transformer coupling and/or magnetic tape recording do).
 
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Thanks Bruce,
it is counterintuitive to most peoples ideas but at higher rate PCM, DSD definitely has its pros as a streaming/archive format over PCM.
Comes down to 1-bit vs 24-bit combined with sampling rate of PCM.
And then considering true transparency of 32/352.8 for PCM (when dealing with DSD) it really hurts as you mention :)

Cheers
Orb

The whole 32bit/352 things makes my head hurt. Is there music offered that takes advantage of 32 bit or is it just a DAC's design capabilities. Just trying to learn a few things.
 
The whole 32bit/352 things makes my head hurt. Is there music offered that takes advantage of 32 bit or is it just a DAC's design capabilities. Just trying to learn a few things.

None that I know of. I use 32bit in my processing and math but nothing in production that I know of. There are no 32-bit capable A-D's and software that will capture it AFAIK....
 
Aren't the 2L download files DXD? 2L says they record in DXD, but surely the DAD ADC's they use aren't 32 bit though. And of course they have the decimation issue that any PCM recording has having been converted to PCM internally from a A/D sigma-Delta modulator front end, like the DAD A/D converter. The exception are recordings made with the Pacific Microsonics ADC, but that's 192KHz tops.
 
Aren't the 2L download files DXD? 2L says they record in DXD, but surely the DAD ADC's they use aren't 32 bit though. And of course they have the decimation issue that any PCM recording has having been converted to PCM internally from a A/D sigma-Delta modulator front end, like the DAD A/D converter. The exception are recordings made with the Pacific Microsonics ADC, but that's 192KHz tops.

Yes, they have the same DAD AX24 as we do and it's only capable of 24bit. What I don't agree with is they do DXD recordings and then upsample to DSD for their SACD's. If it were me, I'd record in DSD first and then go to DXD if needed.
 
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None that I know of. I use 32bit in my processing and math but nothing in production that I know of. There are no 32-bit capable A-D's and software that will capture it AFAIK....

Just for studio processing/DAW isn't it Bruce rather than actual consumer end product.
Cheers
Orb
 
Bruce,
how did the MSB Tech Studio ADC (with 32-bit) compare to Grimm AD1 (DSD)?
I appreciate both have their places in a studio and not really like-for-like.
Cheers
Orb
 
Make PCM sound like DSD


I finally read it. Lynn, nice. I hope to have some useful comments. Notice though that while I actually use your own (said) base for all, I put it it in the 180 degree opposite context. Could be nice ...

Let me first virtually quote from others (like Mani has done it more in the beginning of this thread) that when DSD is compared to PCM, and then in the perceivedbly most direct way (chains) for both the DSD and PCM situation - and which would be self recorded (Tascam) DSD played back through a Mytek vs self recorded (PMII) through the NOS1, DSD sounds saltless. We could say "more analog" but in the end we maybe must redifine how analog is sounding, knowing that the dynamics in there can't be the very best, plus knowing that theoretically digital performs better there. Much more to say with pros and cons for both, but the grasp of it is hopefully clear. So, this PCM wins "hands down" and it is in the dynamic character of the sound, which at the same time is smooth and silky. So, notice the explicit recognition of being dynamic and grainy, which is not difficult to achieve. Have it silke and dynamic, is. And before we forget, this is actually comparing Redbook to DSD, or IOW lowres to hires. This is important knowledge for the below. Now from your article :

The classical PCM spectra has a very different effect on slew-prone analog electronics. The extremely fast rise-times – in the nanoseconds – slews the analog electronics at the rising and falling edge of every sample, and the rise time of the switch array in the converter is pretty much the same whether the converter is operating at 44.1kHz or a much higher frequency like 705.6kHz. The slew events are so short that it doesn't appear on FFT-based distortion measurements, since the FFT measurement is averaged over a second or longer; obviously, a few nanoseconds occupy only a very tiny portion of a second.

Emphasis is mine.
What you assume here, is a high frequency, which at least with Redbook can not be the case (ehm, that much); it is just not in there. Now :
What *does* happen, and in fact all the time, is the very same fast rise time, but from transients. A transient is not a frequency (unless we observe it in the electrical domain) but it is just a one-shot fast rise (never drop) which occurs at attacks (think the rim of a drum for example).
No real world attack rises inifinitely fast (but let's not forget synths), but, because of the too low sample rate it will look like close to that in digital. So, say that this rim shot evolves so fast to its (sort of) sustain level, that from one sample to the other it goes up from 0V to 1V. This really happens 1000s of times (with the 1V as the real example were it for 2VRMS output of a DAC) and thus it has to be dealt with. Has to be ?
Yes, because a transient is a transient and nothing tells that it can't be dealt with. Ok, the slew rate does. Aha. But anyway, because it is not a frequency, it just can be done by any analog device which can follow it (you may recall my dirac pulses plot).

since the sample duration has been affected by the slewing, which is not the same as linear low-pass filtering.

Here you actually touched my subject above. Well, this is how low pass filtering will flatten transients, unjustiffied.

Now your PCM=DSD solution ("BUD");
Although I didn't do the math for merits, it is clear that the DSD principle allows for transients which are derived from the sample speed, the maximum allowed voltage and thus the shortest time possible to reach our 1V from above. Of course it is clear that many samples are needed to get from 0V to 1V and it is only that the e.g. 2.8MHz sample rate can do this fairly fast (notice that at least for SACD it is not allowed to have more than (IIRC) 14 subsequent DC rises so that also slows down). Still PCM does this infinitely fast. And now the fun :

Where you actually slowed down (smoothened) the rise time (hey, and fall time because you talk about frequencies) so it will sound more analogue (actually : more DSD), I can use the very same principe of DSD to point out that it can't deal with transients as good as PCM can.
So ... you knew this, but we both used a different context and I see the infinite possible rise time of PCM as a virtue while you see the slow rise time of DSD as a virtue.

Maybe a year or so back I talked about a similar subject and how the higher frequencies may not be important in the first place, but transients always are, and there it was the first time I could find a reason why PCM-hires sounds worse than Redbook. Please keep in mind, this is "our" NOS1 usage and ever and always knowing that the thing operates electrically 100% the same for either format (because all is done in software preceeding it). So, read your own article Lynn, and see how it can happen indeed that PCM-Hires needs your DSD approach. Why ? because *then* you talk about frequencies and possible messy slewing.

That's it. Shoot me if needed. :cool:
Peter
 
Just for studio processing/DAW isn't it Bruce rather than actual consumer end product.
Cheers
Orb

Correct.... processing only

Bruce,
how did the MSB Tech Studio ADC (with 32-bit) compare to Grimm AD1 (DSD)?
I appreciate both have their places in a studio and not really like-for-like.
Cheers
Orb

I feel the Grimm is the best A/D converter for DSD anywhere. While the MSB is great in DSD, it excels on PCM material, especially 352.8/384kHz.
 
DSD64fs is the equivilent file size of 24/88.2, so I can not agree with your statement. I worked in DXD when I was doing FIM mastering and just 1 track would routinely be over 1GB! An album would be 12-15GB whereas same SACD would be 3-4GB.
Now when you start working at 32/352.8, then you will need RAID arrays and 100's of GB!!

Let's run the numbers, Bruce:
(For a stereo stream- not including parity/redundant bits.)

44.1/16 = 1.4112 MB/sec
88.2/24 = 4.2336 MB/sec (easily as good as DSD)
96/24 = 4.608 MB/sec
64FS DSD = 5.6448 MB/sec
176.4/24 = 8.4672 MB/sec
192/24 = 9.216 MB/sec
128FS DSD = 11.896 MB/sec (a fair comparision to hi-res PCM)
352.8/32 = 22.5792 MB/sec (extravagent by all means but only necessary for computation, not storage.)

Cheers,
Rene'
 
352.8/32 = 22.5792 MB/sec (extravagent by all means but only necessary for computation, not storage.)

Off the subject, but I'm interested in how you arrive at this conclusion. I could demonstrate to you, as I believe Bruce could also, the sound quality difference between a raw 64fs DSD recording of an acoustic space analog event, recorded through the Grimm AD-1, compared to the DXD (352.8/32) reduction of the same file. Going to any lower sampling rate PCM only increases the perceived difference. There's no comparison to 88.2/24 PCM, let alone "easily as good as DSD" IMO.
 
Ragnar,
Yeah DSD becomes more efficient just beyond 96khz/24-bit, but we are talking studio rather than end-product and in that context quite a few will say DSD is at least comparable to 192kHz/24-bit and if wanting transparency working from DSD to PCM then it requires comparable DXD (according to some notable engineers).
Appreciate you were just expanding upon Bruce's post.

BTW quite a few still show you only need 64fs DSD (that includes Bruno Putzey), and equal transparency if wanting to work with that in PCM requires 352kHz/32bit as mentioned earlier.

However in studio it seems practical sense to stream/store as DSD and then work with it in the studio processes/DAW at 352/32bit.
DSD is the ideal for streaming/storage when considering studio requirements, this does not mean it is necessarily the best for consumer DAC playback though.
Cheers
Orb
 

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