[Alex Altorfer]

Marios,

I've decided not to take the SACD route and I didn't buy a 3000 dollar cdp either. Fortunatelly, my player of choice sounds warm, natural, and very musical. There's no digital harshness to my ears. Surely, a 3000 dollar cdp oughta sound a quite a bit better than the Planet 2000 though.

Cheers,
Alex

 

[jefemeister]

Quote:

Originally posted by markl Doesn't this defeat the whole purpose of SACD?

do you mean to imply that there is a point to DSD?

Quote:

Wouldn't you just be better off with a 24/96 DVD-Audio disc on a machine that did that, so it doesn't have to go through that conversion at all, it could stay native 24/96?

I definitely prefer DVD-A. In fact they only problem I have with DVD-A at all is that it is multichannel. I'm a firm believer that music in more than two channels is a travesty. Well that and the fact that you need a TV monitor and have to navigate menus just to get a song to play.

Quote:

Complicating this factor is that I've also read that the vast majority of today's DAC chips are not really multi-bit, but start the process as a bitstream, reading the PCM data off the CD in a single-bit manner a la DSD (SACD), *then* convert it to multi-bit PCM.

I'm not sure this is equivalent. The data is still PCM, as the bits still represent absolute samples instead of DSD's relative sampling scheme. I imagine you mean that the data undergoes a parallel to serial transformation which is really just a shift register. To go to DSD you would need a comparator in there somewhere as well. I don't claim that 1-bit converters don't exist. They were quite popular years ago when CD first came out but have since been demonstrated to be vastly inferior to multi-bit converters.

Quote:

Yes, here's one of the chief "complaints" about SACD that is fodder over at aa. However, this really only comes into play for modern recordings made in a modern studio, not to old analog tapes from analog studio equipment.

I'm not sure I follow you. An analog tape is no different than a singer in a booth. It still has to get from analog to DSD somehow.

Quote:

No way to "add back" information in a DSD conversion that was never recorded in the first place via PCM. That music will never sound "better" than 24/96 data will allow.

agreed.

Quote:

Also, I've read that SACD is not as manipulatable as PCM in the studio, much harder to edit with or to process. It seems to be a combination of lack of equipment combined with technical problems with trying to process a DSD signal.

true. this is something I didn't think about at first. DSD is a relative sampling method in which all current samples are meaningless unless you know exactly what came before them (in DSP terms we call this a non Linear Time Invariant system or LTI.) Most DSP type things break down if the signal is not LTI. Therefore most post processing of the DSD would be impossible. I've never read this, so I kind of went out on a limb with that one but it makes sense. I know that digital volume control can not be applied to DSD because of its relative nature and that 1 or 0 times x is either 1 or 0 and nothing in between.

Quote:

Actually Sony developed SACD as a means of protecting its most valuable asset-- namely aging master tapes that were crumbling before their very eyes.

that and that 1983 + 17 = 2000.

Thanks Mark. This has been the best exchange I've had on head-fi yet.

Jeff

 

[markl]

Here's what Meitner (I'm sure you know of him jefemeister, but for others, he's a digital uber-guru, maker of professional gear and very expensive audiophile gear) said about modern PCM DACs that I clumsily tried to paraphrase (these are 2 separate quotes): Quote:

Meitner: … well, if you have it in the one bit DSD format, you can, (A) you have a pretty robust storage that way, (B) you can now convert it to any other format that may come about, PCM 96/24, whatever. So it’s a very versatile format to begin with. And don’t forget that every A to D converter that you see on the market today starts off as a DSD modulator. So then you have the DSD signal on the A to D that just goes to the PCM down sampler or decimator and gets turned into PCM, so the life of the audio in the digital world really starts off as a one-bit signal. Meitner: Yes. And you know the funny thing is, on top of it, you know all the converter people who started off in the multi-bit scene, have all changed to single bit. Phillips with their bitstream; and look at the vendors of DAC chips and A to D chips and it’s all gone to single bit. You would be hard-pressed today to still find multi-bit converters because of the added problems of them not being enabled to do zero crossing and stuff like that properly.

Here's a link to the whole article from Positive Feedback, 90% of this goes over my head: http://www.positive-feedback.com/pfb...r.rev.8n2.html
Quote:

I definitely prefer DVD-A.

Speaking as an engineer of digital gear or as an audiophile/music lover? How many SACDs have you listened to, on how many pieces of gear? Not an attack at all, just curious how you reached this conclusion. I have around 30-40 SACDs, and about a dozen DVD-As. I've owned 2 DVD-A players and 2 SACDPs ($1000-$1800 range). At this stage, I don't feel I have nearly enough info to pick a favorite format yet. I do know that I like them both over the CD. I have noticed what I believe is a trend in the way they sound, SACDs tend to be much more "analog-y" and natural whereas DVD-As tend to be uber-digital, very hi-rez sounding, more impressive at first, but maybe not as "real". Quote:

In fact they only problem I have with DVD-A at all is that it is multichannel. I'm a firm believer that music in more than two channels is a travesty.

I'm sure that's what they said about 2-channel stereo. "Two channels???? What do I need two channels for? It's all a scheme to force me to buy more amplification and speakers!" In addition to being a music-lover, I'm a movie-lover as well, with a pretty good HT rig, so I've grown accustomed to surround. I sincerely enjoy the surround-sound experience. I also enjoy the multi-channel music experience. No, these recordings don't sound the same as the more familiar stereo versions, but if I'd heard them first in MC, would the stereo version sound "wrong" to me, or collaped, compressed, compacted together and muddled? Then again, a lot of the music I like, is very carefully mixed, multi-tracked with lots of cool stereo panning and phasing effects, it lends itself to surround sound. Turn these geniuses at the mixing desk loose on 5.1 channels and look out! Think what they can do with all those extra channels. Think of future masterpieces, equivalents to Dark Side of the Moon that will be created natively in 5.1, conceived originally as a MC experience, by people who think in 5.1, not stereo. What an amzing experience that will be! I think ultimately, the future of music is multi-channel. Maybe far into the future (20-30 years), but it is coming.

Also, with all those additional channels, you get 3X the resolution you than you can get by compacting all those tracks into two channels. Sounds can have greater separation, and more ambience, bigger, wider soundstages, etc. OK, you get my point, I think MC music can be wonderful. We're only just now learning how to mix music into 5.1. Think of how bad and funny early stereo recordings sound to us today, it took them a couple decades to get it right.
Quote:

I'm not sure I follow you. An analog tape is no different than a singer in a booth. It still has to get from analog to DSD somehow.

Yes, it is still a conversion of analog to digital. But a DSD copy of a 24/96 recording can only ever have 24/96 resolution. A DSD version of an analog master tape is a much closer approximation to analog with more samples/information than 24/96 PCM. Again, Sony developed SACD specifically to best preserve aging, brittle analog tapes before they disintegrated. They weren't thinking about new, modern 24/96 recordings when they came up with the format. Here again is Meitner: Quote:

Meitner: To convert audio into PCM is a very alien thing, whereas if you look at the convert audio into one-bit format, it’s a very natural thing. In any form of conversion, you will lose something. You have to choose the format where you lose the least, which means the format that’s the friendliest to audio, which is definitely DSD over PCM. For sound quality, here’s one simple test. I’m doing some transfers of vinyl LPs onto DSD. And, you know, in DSD this is a conversion with a minimal amount of damage to the original sound. If you consider playing back vinyl and liking all the good things about it, now we can have it in DSD format. We could possibly take some of the clicks and pops out of it and still have the general good flavor preserved. The same holds true for analogue tapes and any kind of conversion. So it’s really a very nice thing.

 

[Alex Altorfer]

Jefemeister and Markl,

Your exchange is intriguing. Please go on. Most enlightening, really. You got me thinking about multichannel audio differently. Maybe someday we'll have multichannel or surround sound cans that don't compromise on sound quality.

Cheers,
Alex

 

[jefemeister]

I haven't given the Meitner link a look yet but look forward to it. I won't comment on A/D converter topologies and modulators because, quite frankly, he knows them a lot better than I. But something seems strange right off that bat:

Quote:

if you have it in the one bit DSD format, you can, (A) you have a pretty robust storage that way, (B) you can now convert it to any other format that may come about, PCM 96/24, whatever. So it’s a very versatile format to begin with.

I don't see how this is different than storing the data in PCM from the start and then converting to other formats. DSD can be created by looking at the PCM sample by sample. If the next sample in line is grater than the previous, then the DSD bit is a 1. If it is less than the previous then the sample is a 0. This is much simpler than going the other way. However this method of PCM->DSD will not increse the sample rate. But if you first upsample the PCM 64x (for redbook anyway) and then do the conversion you could hit the 2.8 MHz DSD sample rate. The upsample wouldn't even have to be very good. I imagine linear would work fine since we're only interested here in wheter things are bigger or smaller, not absolutes. [edit: on a 2nd reading, I see that last sentance about linear technique is not true. ignore it. You have to do a for real upsample but that's pretty easy anyway] Going from 44.1/16 to 24/96 is a different story though and maybe that's what he's refering to specfically. We've discussed earlier how DSD isn't very versatile when it comes to mastering--such as controlling scale, reverb, compression, etc. I don't know for sure but like I said I don't think those things can be done with DSD.

Quote:

You would be hard-pressed today to still find multi-bit converters because of the added problems of them not being enabled to do zero crossing and stuff like that properly

I don't mean to refute him here because, again, I think he's got the knowledge/experience jump on me. But multibit style converters really excel at low level detail. That is they have much higher resolution than their competition when the signal is at it's lowest amplitudes. Music signals are most often right around the zero point. If you look at a wav file you will notice that the vast majority of the time the amplitude is hovering right around the zero point. Of course this means that a lot of "zero crossing" also occurs. To be honest I'm not quite sure how that effects performance in any type of DAC without making some large asumptions on my part. so I won't.

Back to Mark's thoughts.

Quote:

Speaking as an engineer of digital gear or as an audiophile/music lover?

To be honest. As an engineer. I have much more experience with DVD-A then SACD but a million times more experience in redbook. I also do most of my redbook listening on a $60,000+ system in a dedicated, acoustically treated listening room whereas most of my hi-rez experience in comparatively substandard environments. Not that I don't hear redbook in those environments too, but I am very honed in on what redbook is *capable* of. That's why I'm usually careful to not discount SACD completely. I throw a few jabs here and there but that's just because I really believe it is a fundamentally flawed format. DVD-A seems more robust of a format. I agree with (and perhaps am more used to) the digital concepts assocaited with it. Remember, I have the disclaimer of being a DSP/architecture guy and we always think of things as registers, buses, FIR fiters, etc which are all multibit type structures. I'm digging myself into a hole here so on to the next point.

Quote:

I'm sure that's what they said about 2-channel stereo. "Two channels???? What do I need two channels for? It's all a scheme to force me to buy more amplification and speakers!"

that is true actually.

Quote:

I'm a movie-lover as well, with a pretty good HT rig, so I've grown accustomed to surround. I sincerely enjoy the surround-sound experience.

Me too. for movies it's fantastic. Especially if things blow up and the bass practically knocks you over, etc. But I wouldn't want to watch Casablanca or some kind of a low key flick in surround.

Quote:

Turn these geniuses at the mixing desk loose on 5.1 channels and look out! Think what they can do with all those extra channels.

That's exactly what I'm afraid of. You and I and everyone else here know that the *vast* majority of mixers as no geniuses. (Yes, it's probably the label owners to an extent too.) There is nary a CD that gets released these days that doesn't sound like crap and that's only 2 channels. on top of that I've got drum kits where the cymbals are full on the right channel and the kick drum full on the left. That's no way to mix music. I can just imagine the damn drum kit flying around my head in multi-channel. That may be cool in electronic or something but it unacceptable in any other genre. If you could guarantee that Bob Ludwig does all the multichannel discs then I think it could have merit.

Quote:

Also, with all those additional channels, you get 3X the resolution you than you can get by compacting all those tracks into two channels. Sounds can have greater separation, and more ambience, bigger, wider soundstages, etc.

I don't see that at all. The soundstaging part, yes. But that's all artificial due to multi channels. It doesn't have anything to do with the inherent quality if the disc/recording. You also are going to have an insane number of room reflections in which you'll have total signal dropout at some nodes and 5x the volume at others. You actually get less resolution from a data standpoint because you have at least 5x the data to put on the disc now. There will actually be more crossover distortion and channel seperation issues because you are dealing with more channels, more electrical components, etc.

Quote:

Yes, it is still a conversion of analog to digital. But a DSD copy of a 24/96 recording can only ever have 24/96 resolution. A DSD version of an analog master tape is a much closer approximation to analog with more samples/information than 24/96 PCM.

I thought you were refering to analog master tapes originally. It is true that if you look at a DSD "waveform" that it does closely approximate the analog signal. But if you did PCM at 2.8 MHz it would as well. Looking at 44.1kHz doesn't seem the same because you have just over two samples/period. So one sample can be positive, the next poitive, but inbetween the original analog actually went negative. But Nyquist says that it doesn't matter because of the nature of sine waves and superpositioning. I really don't buy into the whole "you can see the analog waveform in the stream" argument.

Jeff

 

[marios_mar]

I dont think that comparing SACD to CD is like comparing two different fruits. They are to formats one replacing the other (at least for audiophiles.

So if one can have an expensive CDP that has warm and full sound that satisfies why bother buying the expensive and of lower variety SACD discs. Anyone who gets a $300 SACDp should rather get a $1000 CDP or something more expensive right?



Now a tougher call
$700 SACD or $2000 CDP (smaller gap)


Doesn anyone know what kind of encoding takes place in SACD multichannel?
And on DVD-A is it 44/16 you got on multichannel recordings?



Still can anyone tell me will the $300 SACD have ANYTHING absolutely ANY good quality that reaches or surpasses the one of the CDP?
I am not into buying either of those but I am just wondering.
Thanks

 

[Uncledan]

To me, SACD and CD are TWO different thing, they gives TWO different sounds. I really love SACD, however, sometime SACD give TOO detail and clean sound, which I don't like in some records.

BTW, a expensive CDP sound much much better then low-end CDP. I don't believe this before, but when you have a chance to listen to them side by side. You WILL understand what I mean.

 

[Wodgy]

A couple of points that haven't been mentioned:

Wadia's DACs convert DSD to PCM internally because their proprietary spline interpolation scheme simply doesn't work on raw 1-bit data. It's conceivable that a next-generation spline interpolator could be derived for running DSD data, but my gut tells me that this would have to use higher-order splines (i.e. more computationally intensive) than their current system, because low-order splines have nasty properties when used for extrapolation.

Meitner is not entirely correct in his argument that DSD is a natural match for analog to digital conversion. It's been shown (by Lip****z et. al.) that in order to generate a good single-bit (i.e. DSD) output from an ADC, you need to use a multibit scheme within the ADC. So it's not clear that DSD is any more of a natural match for analog to digital conversion than PCM.

 

[kentamcolin]

For me it will just be nice when SACD becomes the norm, if ever. It's not certain right now that it will, and there are very limited titles. Anyone remember movies on those LP size laserdiscs? When SACD is common, then it will be worth my investing in a good player, or a player at all. Some can afford this for the few titles out now, and that's great for them. Most of us are not in that boat. I also don't really want racks and racks of gear. I already have three formats; vinyl, CD, radio. I like to keep things as simple as possible, my personal preference.

I also don't get the point of arguing price-points. I recently sold an integrated tube amp (Cayin TA-30). While it was listed, someone emailed me and asked if it would sound as good as a certain $6K amp he recently auditioned. He couldn't afford the $6K but was looking for that sound. Duh! If you have $3K to spend on a source, why would you buy a $300 SACD player? How about a $3K universal player? If you only have $300 to spend, why buy an SACD player (even if it did sound that good) when there is such a limited selection of music? I guess I'm ranting a bit, but so often we hear of gear at a certain price that is supposed to sound as good as anything twice the price. But was it compared to the best at twice the price?

Maybe I'm just a little uptight this morning

 

[Canman]

Quote:

Originally posted by kentamcolin Duh! If you have $3K to spend on a source, why would you buy a $300 SACD player? How about a $3K universal player? If you only have $300 to spend, why buy an SACD player (even if it did sound that good) when there is such a limited selection of music?

kentamcolin, thank you for saying what should have been said long ago.

markl and jefemeister: thanks for some interesting reading.


Quote:

Originally posted by marklYes, it is still a conversion of analog to digital. But a DSD copy of a 24/96 recording can only ever have 24/96 resolution. A DSD version of an analog master tape is a much closer approximation to analog with more samples/information than 24/96 PCM.

Markl, while it is true that DSD contains more samples than PCM, does it necessarily contain more information / data than 24/96?

 

[markl]

Quote:

To be honest. As an engineer. I have much more experience with DVD-A then SACD but a million times more experience in redbook. I also do most of my redbook listening on a $60,000+ system in a dedicated, acoustically treated listening room whereas most of my hi-rez experience in comparatively substandard environments. Not that I don't hear redbook in those environments too, but I am very honed in on what redbook is *capable* of. That's why I'm usually careful to not discount SACD completely. I throw a few jabs here and there but that's just because I really believe it is a fundamentally flawed format.

Uh-oh, the "SACD is fundamnetally flawed" argument rears its ugly head at Head-Fi!

Here's a now legendary (and HUGE) thread from mastering Maestro Steve Hoffman's web site, "Is SACD fundamentally flawed": http://www.stevehoffman.tv/forums/sh...threadid=26075

This argument has to do with the potential issue of the introduction of high-frequency noise in SACD that may or may not (depending on who you ask and phase of the moon) intrude on the audible range of the upper frequencies of SACD discs. The fear is that the 1-bit scheme has some troubles above 10KHz or so, that can actually reduce resolution at those frequencies, despite the fact that SACD claims much expanded frequency response over the Redbook CD. Over at audioasylum they debate endlessly whether this noise exists, and if it does, how it *might* affect sound. Some claim that this distortion is the reason that some SACDs can sound "soft" to *some* people in the highs. My feeling is that this perceived "softness" is actually due to the absence of the harsh PCM signature, and more analog-like sound of SACD, but who knows. Some assert that newer converters have solved this problem, though. I wish I knew the full story, but that will require much more research time than I have. Stereophile's position is that this distortion *might* be there, but even if it is, it won't be perceptible by listeners.

It is important to note that for every "DSD/SACD is fundamentally flawed and here's why" argument that can be made there is an equally damning "PCM/DVD-A is fundamentally flawed" rebuttal.

Further controversy about SACD was touched off by one Bruno Putzeys, a chief digital engineer at Phillips who is no fan of SACD and spoke ill of the format out of class on an audio board (I have spliced together the most interesting points he makes here, Putzeys actually reples directly to many members questions toward the end). Quote:

"SACD is indeed fundamentally flawed. Using 1-bit as a conversion method can be a valid choice when the analog circuit does not have performance higher than the 1-bit signal. To use this as a data format, thus binding everyone to the noise and distortion limits, is quite another thing... SACD is a typically Japannish invention in that it is a solution to a nonexistent problem (decimation-interpolation), which in turn creates some very real problems left for real engineers to solve. Some examples: 1. Splicing (editing) two DSD signals together creates a "click", even if both represent silence. 2. Any processing (except delay results in a longer word length. Getting back to 1-bit requires another stage of deltasigma modulation. Sony dreamt of a new signal processing paradigm operating entirely in DSD. It was not to be - they even officially admit it now. Any quantisation mixes the signal with quantisation noise. They can no longer be separated. This is not much of a problem at 24 bits. At 1 bit however... well... 3. The accumulated noise from previous conversions reduces the deltasigma modulator's headroom. After 5 conversions (e.g. level control, eq, mixing, fader etc), the modulator already overloads at silence. 4. DSD is not distortion-free. 5. The signal bandwidth and the noise zone overlap. In a correctly designed converter, the signal occupies the "clean zone" only, thus allowing the noise to be filtered away. With DSD, the noise zone starts at 20kHz but the signal bandwidth extends -by Sony's definition- to 100kHz. The SNR over 100kHz is only 30dB. Many amplifiers produce audible distortions when presented with this noise (hence the switchable filter on many SACDs). It was "invented" when someone took a CS5390 chip, wired the 1-bit test outputs straight to a D/A converter and liked what he heard. Thus, the standard was fixed at 1-bit/64fs which happened to be the internal operating parameters of this particular chip. As far as I know, all comparisons between DSD and PCM have been flawed in one way or another. Even in players where the same DAC chip handles both formats, the converter is run at higher speed using a different modulation scheme. Comparisons between both formats, for the better or the worse, get severely skewed. I've recently built a converter that converts PCM and DSD using the same signal path. That is, the PCM first gets converted to a signal per DSD specs and this signal is then converted to analog. In this way the only difference between the two modes is that in PCM mode, the audio content is band-limited to 20kHz (plus possibly some minor aliases, see "halfband") and the quantisation noise floor in the audio band is at the 16 bit level, whereas the DSD -3dB bandwidth is approximately 70kHz and the quantisation noise floor in the audio band is at approximately 20 bit level. The outband noise is pretty equal in both cases (ie shooting up fearsomely after 20kHz). It affords a good comparison between low and hi rez, as the difference lies precisely in bandwidth and noise floor. Well, I daresay I'm a bit underwhelmed by the difference. OK, it is quite noticeable and if I get the choice, I prefer the high-res playback, but to proclaim DSD or 192/24 "da bomb" is IMHO an extreme case of hyperbole. Apart from this, I'd like to attract attention to the fact that the majority of SACDs actually contain no more than the original digital master (48kHz or even 44.1kHz) converted to DSD. Would you care to elaborate on the implications of a > high-ranking Philips engineer taking this stance? It's no longer a secret. I can do this. ( Before it was out in the open, I used to be quite vocal on this issue as well though. The result would be my colleagues going "Shhhhh!", followed by a burst of laughter from everyone. ) From quite early on, Philips' position on SACD has been to emphasize on the multichannel capability (6 full-bandwidth channels available on a full-length album), not so much on the DSD scheme.

TH:
Studio people who have compared the live misc feed to DSD and PCM say that DSD is much better and they cannot tell the two apart. If, as you say, it is flawed, why are the studio engineers all for it. As someone on the forum said, it is analog without all the problems.

BP:
1) Sound of DSD:

The HF noise and the low-level nonlinearities of DSD do not get in the way of the sound as we hear it. However, it is impossible to build a production chain using the format.

DSD is in its place in one (1) application: Mastering from analogue. When the recording chain is completely analogue, you can feed the audio from the analogue mastering into a DSD A/D converter and cut that signal straight onto a disc without any further processing. It is in this application that DSD can be viewed as pretty transparent. When you convert the signal twice, however (such as when using a DSD recorder as the tracking medium), the second conversion is no longer transparent, due to the HF noise present in the source signal hitting a second analogue deltasigma modulator.

In this I have a serious gripe with AudioQuest. Before DSD, they tracked onto a 15ips 1/2" two track. These tapes were first used to master to ordinary CD, later again to SACD for a reissue, which indeed was a sonic forward step. However, they then switched to using a DSD recorder for tracking as well. Grundman then mastered it as usual (ie through an analogue mastering studio) onto the DSD master recorder.
The recording guys may well have found the DSD recorder better than their analogue two-track in a dry shoot-out, so you can't blame them for having made that choice. The resulting SACD releases however, are below anything. What's worse, there's no way of salvaging the recordings for a better-sounding reissue (unless they had the analogue two-track running as a backup). As an example, get hold of the classic "BluesQuest" sacd, which was made from analogue tapes. Then compare Doug McLeod's "Whose thuth whose lies", where a DSD machine was used. If you're a bit of a sensitive person you'll run away screaming.

I think we could say that DSD is analog with a few extra problems. Serious ones.

2) Total transparency:

Apart from that I consider any claim of "not being able to discern between a live feed and DSD" as something of a hyperbole. While flicking a switch during the music will indeed not reveal any serious deficiencies, more controlled listening (e.g. listen - rewind - switch - listen) will certainly get you hearing a lot more. I can't even insert a unity gain stage (low-noise low-distortion etc) in the signal path without hearing degradation, let alone a pair of converters.

3) Jitter:

When a signal is fed straight from an ADC into a DAC, they share the clock. When you sample a signal with some jitter, and you reproduce it with the *same* jitter, the jitter has NO impact on the sound (jitter which impacts the sound most, incidentally is of the LF kind, which does not affect noise specs, and which is typically not attenuated by PLLs). With DSD this works - the delay between ADC and DAC is nearly zero. With PCM it doesn't, because there is a delay of several milliseconds between the two, meaning sampling and reconstruction see different jitter. This means that a live vs DSD will always sound more transparent than when you take the digital signal to tape and replay it.

4) PCM implementation issues:

On www.nanophon.com you can read a number of the late Julian Dunn's excellent papers on how compromised implementation of digital filters account for many of the deficiencies noted with PCM. These can be readily solved with due care for details and at the expense of only mildly increased computational burden.


TH:
How do you feel about the DSD workstations? The existing ones, I believe, are all using DSD-wide, or PCM-narrow.

BP:
The workstation from Merging is in itself OK in that the data is kept in 352.8kHz/32bit floating until it is written to an sacd master file. Currently the incoming data is DSD, causing a "two DSD conversions" problem (although apparently in the digital domain the sonic degradation is less than when it happens in the analogue domain). If you can somehow get the data in through a non-DSD converter (preferably at 352.8kHz/24bit or so), it would become a digital equivalent of the "ideal analogue chain" in that only one deltasigma coder is in the signal path (when writing the master). That would be fine.

If you do have to edit a recording which is already in DSD, it can be done in a sonically transparent way by using the "transport" mode of the workstation. In this mode, the DSD data is preserved exactly except during the edits (crossfades). The two conversions problem would be restricted to those edits only. Of course, any form of processing, such as level change, EQ or mixing, is not possible in this mode. It's really just cutting and splicing.

The workstations from the Sony camp (I don't know names) use 2.8224MHz, 8 bits, still noise shaped. This still has the same problem as reconverting to DSD every time, but reduced by 48dB. Still I wouldn't recommend it. The Merging approach is the most suitable one.


TH:
You mention the use of a non-DSD converter (preferably at 352.8kHz/24bit or so). EE Dan Lavry argues that the bigger the numbers the less accurate the conversion is. He says that there really is no need to go beyond 24/96 (48k bandwidth) and anything more will not result in a more accurate signal but rather more noise and distortion.

BP:
Dan is correct in that as the sampling frequency is increased, the available signal to noise ratio inside the nyquist band decreases. However, when you keep the bandwidth across which you are measuring SNR constant (e.g. you measure noise across 20kHz) and when noise shaping is used, this trend is often reversed. DSD is precisely a case in point. I have one converter here (homegrown discrete circuit) that puts out 1 bit at 2.8224MHz. Measured across its nyquist bandwidth (1.4112MHz), its SNR is useless, well below 6dB. However, taken across 20kHz, it delivers the full 120dB. When you push the sampling rate too high, performance will again resume a downward trend.

Dan's converters are multibit, non-noise shaping. Such converters will not even hold their precision at constant bandwidth when sampling rate is increased. Since his converters have ultra-low noise as their hallmark, he's quite right to maintain a reasonable sampling rate.

When I propose to use a 352.8kHz converter, in practice that would be a noise shaped converter designed to offer maximum SNR performance over as wide as possible a band, but not up to 176.4kHz (not feasible). It would still have a "noise shaping tail" in the nyquist band, but much less so than DSD. At the final DSD conversion stage, the noise of the ADC would be negligible compared to the DSD noise, so the output spectrum would be pretty much the same as that of an analog signal converted to DSD in one go.

A design which is in the work at home uses 64fs, 16level PWM with 7th order noise shaping. This would offer up to 129dB SNR (limited by amplifier noise) over 80kHz, which is 4 times as wide as DSD. This would insure all the flexibility of PCM while a later conversion to DSD is not compromised.


TH:
How do you feel about DSD as an archiving format? Is the transparency good enough for Sony's precious analog master tapes? And what makes DSD ideal for archiving as opposed to PCM? Or does it matter? I read Neil Young chose hi-res PCM format for his masters.

BP:
When sony came up with DSD as an archiving system there was hardly even 96kHz PCM around. If at that time they had some old tapes to archive before they fell apart, DSD was the best available. However, since DSD is a liability in terms of processability, archiving to DSD now is no longer a good idea and use of 192/24 is warranted instead. Since SACD is probably here to stay we should view DSD as strictly a release format, in the same way as we didn't produce on vinyl, but music was brought to the home on it.


TH:
What do you say when you hear audiophiles make comments that DSD has all the "air" and "smoothness" of analog?

BP:
In my own experience, high speed PCM also produces the air DSD has, while the "digital glare" of some PCM can even be solved at low speeds. It is caused by the narrow alias-band that is present between 20 and 24.1kHz. Removing this band prior to playback restores naturalness and focus.

I don't suspect DSD of any "euphonic" effects, although, who knows, the HF noise?

Admittedly, the DSD camp has been able to mobilise more audiophile designers (folks like Ed Meitner), resulting in the analog circuitry of the best DSD converters sounding better than that of most available PCM converters. Doing a straight shoot-out is actually quite challenging technically, as the two formats normally use different converters, necessarily producing a different sound. This is another reason for me to do this 352.8kHz converter, because its output can be converted to either DSD or 192/24 while compromising the performance of neither. This would finally allow a direct comparison.[/QUOTE]

So, what does all this reveal? To me, as someone who works in technology in product management, what I observe here is a classic dilemma between two organizations that have to work together on a project (in this case Sony & Phillips). One part of the organization (Sony) developed the product, and it's up to the other to adopt their technology and incorporate it into their own (Phillips). Putzeys has a real case of "not invented here" as well as a superiority complex about Phillips engineers (or "the real engineers" as he puts it) vs. those dolts over at Sony who threw this lemon over the wall for our team to "fix".

It should also be noted that the majority of his beefs with SACD do not relate to how it performs sonically for the consumer (he says he thinks it sounds better than CD), but it's impracticality as a digital medium for the creation of music by engineers in the studio (again, it's the editing and processing issues he highglights). But, it's so early in DSD/SACD's evolution that who's to say these problems can't be overcome with a little "real" engineering?

So, I take everything he says with a grain of salt (and a little sour grapes, too).

Need a break, I'll come back.

Mark

 

[stuartr]

Quote:

SACD is a typically Japannish invention in that it is a solution to a nonexistent problem (decimation-interpolation), which in turn creates some very real problems left for real engineers to solve.

For this reason alone, I completely disregard this moron's expectorations. Whether or not he is right, anyone who casts their arguments in such a racist and baseless manner should be smacked upside the head.

 

[kuma]

Quote:

Originally posted by marios_mar How would a $3000 CDP compare to a $200-$300 SACD player?

At 3k$ mark on redbook player SACD playback on 200$ SACD player vs. conventional CD playback, SACD, if the software is well mastered ( they are at the mercy of recording quality, too ) , can sound remarkably good and impressive.


Quote:

I mean wont REDBOOK be always redbook with harshness etc?

Not necessarily. That's a stereotype.

 

[jefemeister]

Quote:

Originally posted by stuartr For this reason alone, I completely disregard this moron's expectorations. Whether or not he is right, anyone who casts their arguments in such a racist and baseless manner should be smacked upside the head.

I'll give him the benefit of the doubt and hope that he's refering to Sony's marketing team when speaking of inventors. Thus the 'real engineers' is refering to making the idea become reality whether they be from Japan or Europe. Who knows though. If not, I'll chalk it up to competition/rivalry between two huge corporations that happen to be on other sides of the world and thus have different colored skin as well.

Mark, I have some comments/reactions that I'll get to later on tonight. Good reading though.

 

[NetRunner]

One thing about SACD and Stereophiles' take in limited bandwith at high frequencies was seen at a their Esoteric DV-50 review I just read.. (Description of the fig. 5 at the bottom.) That is, they did a measurement that, at least to me, makes it look like SACD was somehow inferior to DVD-A and even regular CD at highs (considering noise floor at least). Just a measurement of course... *Meaning it doesn't necessarily relate to the actual sound..*