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It looks to me that MC SACD has a lot of catching up to do, especially in the area of channel management and quality players. Is there any such thing as a native and transparent DSD DSP? PCM is handled very transparently by more than a few affordable players and receivers. As to quality players, DVD-A offers a continuum of complete one and two box solutions from the Pioneer 47Ai to the Meridian 800. And SACD offers....? If I had to do it right now I'd end up with a modded Philips 1000 hooked up to a Meitner DAC hooked up to an Outlaw ICBM!!! Bring on the DVD-A software.
Follow Ups:
DSD DSP probably does more harm than good at least in today's consumer grade products. For this reason, I'm trying to come up with acceptable full-range speaker configurations so I don't have BM issues. Time-alignment is not a problem as I have the space for equidistant speakers.OTOH, not everyone has the luxury of full-range, equidistant, even identical speakers....
Neverminding the sonic differences, DVD-A needs to do a much better job with software selection...I am still waiting...
Are you right? DVD-A does not have significant high end support and very low end software support.
The same applies to the software side of things - a DVD Audio project is much more sophisticated than an SACD cracker... oops... I mean disc.Meridian, Denon, Audionet, Marantz, Samsung offer high-end DVD-Video/Audio players while quite a few have universal solutions.
Yes, SACDs are exploding, cracking, and shattering everywhere. I was waiting for this cheap hit...It's one pressing plant, and one particular formulation, moron.
Not all SACDs, not all Hybrids, and not even all Hybrids from Crest.SACDs are DVD 4.7's, and even you should recognize that there's no difference, inherently, between the CONSTRUCTION of SACDs and DVD-As.
It's all DVD media, champ. Crest presses DVD-A's, too. It isn't an SACD problem, it's a media issue that'll effect both the Axis and the Allies. Just wait until olefine DVD-A's get produced. You'll be putting your foot in your mouth.
Which is fortunate, since I'd like to put mine through your face.
You know full well the same happens in reverse on the hi-way. No need to defend SACD continually. You should also very well know that there are many on the hi-way that will tolerate nothing but positive comments on SACD. It is refreshing to see the discussion on the hi-way of this problem. At least heads are out of the sand. I see no one from the audiobahn over on the hiway crowing about the disc problems and that would be easy to do. I am sure it can be readily and easily dealt with and is not a long term problem.More problematic is the SACD specs. The discs and hardware seem to be much more finicky and to fine tolerances than CD or even DVD. The discs need to be handled very carefully and there have been constant issues with both Sony and Phillips hardware. Marantz seems to have dodged the buggy hardware issue but you must admit, with technology similar to CD, ie. same transports, there have been way more problems with hardware than one would think for a technology that is not brand new. This leads one to surmise that the smaller pits and tighter lazer beam focusing are the problems. This seems to create more problems with dust, dirt and the general handling of the discs that CD has not experienced. Anyway, you should be commended for your fine work in this matter and I view you as one of the few people who knowledgeably comment on both formats without picking favourites. Without getting into a Michi love fest, your view comes with much more respect than many other more partisan in nature. Regards,
...they're kind of embarrassing actually. But it's the hate-fests i'm not too partial to.There are stupid things said on both sides... I mean, I have to watch my back over here, because if I'm not negative enough, I may get static over there when I go back. :)
I feel like I'm working for the mafia and the FBI at the same time. :)
It can be a pretty funny (but serious) issue getting a lot of air play on multiple forums.Of course, DVDAphile often reminds me of that el-Shahif (sp?) character who played the role of Iraqi minister of (dis)information....
Whats this person's problem? Poor upbringing or just sexually frustrated individual?> Which is fortunate, since I'd like to put mine through your face.
SEXUALLY FRUSTATED....CRETIN !
I've dyed blue hair and smoke clove cigarettes. This means I'm an 'angry youth'.As far as sexual frustration goes? Well, no comment on that, but my Ryan Adams SACD is almost as good to look at as it is to listen to.
ahem.
Just wondering, blue hair + japanese.... its a long shot.. but had to ask. Japanese music is something I've just been getting into.
But on a few suggestions, I think i'm going to check that out.
I am gray haired ( if any ) with a big fat belly...AndI like Mozart...Sounds too good to be true ?
For the rest ? Long long time ago....
But, Baby, I am a man*...
right here in the middle of the Audiobahn... Geez, that's not allowed.
:))Best
Comment ca va Eric ?
Yes an moment that will last an life long....
And we can make beautiful music together.
...well......At least play it back.
Just last night you were flaming the place inside out, and today you're on a honeymoon with a vinyl-perfectionist Frenchman..
How unpredictible can you get? :))Best
I honestly don't stay angry long.
er... maybe not...
The perfect match for you...
is a word that goes together well..
How about this for a starter...if your name is...Michelle...
Salut l'ami.. la carpe et le lapin?Things are ok here, but business is in slow motion right now.
- Do you have A la Vie A la Mort SACD yet?
Best
Eric
PS I enjoyed the EMI DVD-A Classics. I'll post a short review.
Yes very bad buisness...Is the J.H. already out ? But I do not think that I will buy it...in fact I never did buy one of his record even if some of his songs were not bad at all----
P.
Yes JH's out, a double album (I haven't seen it yet, but it's advertised now). It might be good, but it might be very bad also, many different songwriters were involved. I only have some of his old songs.Cheers
Can believe that....I will have a listen on Amazon for good old Johnny....And on France Inter I may have already hear some of his new songs...
Salut,
...Not all of us hide behind unregistered monikers.
The same type of idiots who could not tell the difference between the Iraqi National Museum and the Ministry of Oil Affairs, it seems. I have no problem choosing this side of the fence...
...as I did to DSOTM in HiRez.
you're in one of your 'zones' right now and it's not pretty.
Michi,The problem is that it's a highly visible title and I have seen reports that this is NOT limited to the DSoTM release. Another reported title is the Police's Singles title. Neither of my copies of these discs are problematic to date, but I haven't checked the pressing plant they came from.
Also, you have made a comment here, that has no supporting evidence....
"It isn't an SACD problem, it's a media issue that'll effect both the Axis and the Allies. Just wait until olefine DVD-A's get produced. You'll be putting your foot in your mouth."
Bonding techniques for hybrid SACD are different from the proposed DVD flipper. Until hybrid DVD-As hit the market, what you've typed is nothing more or less than idle speculation.
Sleeping with the enemy?This still is NOT SOMETHING THAT IS INHERENT TO THE SACD FORMAT.
IT HAS TO DO WITH A PLASTIC USED AT ONE PRESSING PLANT.
Try this sometime. Look at DVDs and hybrid SACDs. The glue does not go to the center, and there is a gap. This is the "BONDING" you talked about.
These cracks are forming where there is none of this glue. So how can it have anything to do with BONDING DIFFERENCES?
It has to do with the CHOICE OF PLASTICS BY ONE PRESSING PLANT.
"CRACKS" ARE NOT INHERENT TO THE SACD FORMAT.
But as you said, things are backwards here. And what you say becomes truth simply because your DVD-A THUGS are poised in the shadows ready to hit any "auslanders" with a Louisville Slugger because they don't talk the talk and walk the walk.
Michi,You're throwing a lot of broad commentary here that is unsupported and (at best) highly misguided.
I buy recordings on whatever hi-rez option is available, although my format of preference is decidedly DVD-Audio.
That's my pragmatic approach to improving the sonics of my music collection.
With the splitting of the forums, it was kind of inevitable though, wasn't it?So, If I get told "You're on THAT side! Get out of here!", or pick up that notion, I get defensive.
The problem is, you've got to be completely ANTI SACD around here to be able to criticise any *specific* shortcoming in DVD-A, even if it is minutia.
It's this bi-partisan, all or nothing mentality that I see cast on *me* when I come in here that puts me on the defensive when someone says something like "SACDs CRACK." ...
That's like saying "Audio equipment gets hot!" ... That's right. Some of it does.
the boards were split because of the constant harping between "camps". I don't go onto the hi-rez board and bitch about anti-DVD-A comments, its not worth the trouble. But if I did, I would expect that the usual suspects would jump all over me. You basically have to view the forum you are in with respect. I post on the hi-way and I try to be very careful what I post. It is not my job to try and stamp out any anti-DVD-A comments. It's bound to happen so let it be. Take the analogy that you have walked into a Republican convention as a Democrat and started slagging everyone that said anything anti-democrat. How do you think you'll be received? Its not worth the energy. So try to chill and realize that there will be anti-SACD comments here and pro-DVD-A commentary. That's how its structured. It will not change.
...Because as far as this stuff goes, I'm not the equivalent of a "democrat".If I were a moderate at a Republican convention, I'd likely say, "Well, I like this this and this, but I don't agree wiith this."
It's you guys who labeled me "democrat", "outsider", "SACD-dedicate."
That isn't the truth. But it's a "You're completely for us, or completely against us!" mentality.
Saying that *all* SACDs crack is just dumb. If someone said, perhaps, on Hirez, that *all* DVD-A's don't have dedicated stereo, that'd be dumb, too.
They don't all have stereo, only some do. Am I dumb ?
n/t
The only reason you think it was inappropriate or 'hysterical' of me to outline which pressings weren't cracking was because it gave you a chance to attack me because I don't rub against DVD-A displays in stores and dare touch the 'tainted format'.If it was DVD-A that had a problem like this, (and it could happen), you'd be all roses.
"RETURN defective items" ... What, for another defective item?
No, no, I understand. If it's DVD-A, it's golden, if it's SACD, it's fecal.
My posting specifying which discs were and were not defective was perfectly reasonable.
You classifying *ALL* sacd's as "crackers" was propaganda.
But, this is a war. So I guess all is fair. So it's fair for you to smear me because I *DARED* post something fucking informative about the format that your Jihad rages against, it's fair for me to say that I hope that your Warners' "Crash! Bang! Boom!" DVD-A develops a fracture while it's spinning, and shatters in your face and blinds you while you're listening to sound-effects whirl about your head amidst a sea of "hard hittin' bass".
or something for my side of the war? 'Cause I *really* suck at that kinda stuff. SACD good. DVD-A good. 8-track bad. (Hey, there isn't an 8-track forum here somewhere is there?) Oh ****, I just offended somebody in the vintage forum ;)
No offense, really. I just wonder why I am on the receiving end of such harsh criticism of yours... I would call this "hysterical" if I may... Your postings are okay with me, otherwise. And it was good to read in another post that SACD bigmouth Groovenoter himself has some bad pressings out as well...To calm you down:
I have experienced problems with a bad Warner DVD-A pressing once, and the Sonopress hybrid DVD-Audio disc I received for testing did not play the way it should in the Denon A-1, Panasonic RP91/RA71, Harman Kardon DVD 1, my Toshiba DVD-ROM,...
In case you want to try any non-crackative European pressing, be it SACD or DVD-Audio or CD, I am ready to swap these with anyone for any American disc I cannot find here...
and it makes my head spin when others do it.Overview. I do not have ANY problem with DVD-A or PCM. It's been made clear that *I* don't like watermarking because I've WORKED with Verance material and don't like it.
And I CERTAINLY don't like the crack problem with Crest hybrids.
I have posted here trying to defend against the "MLP is flawed" crap, which I don't believe.
But, then, I also post against the "DSD noise is audible" crap which I also don't believe in Hirez.
I address specifics. "DVD-A has this problem", but what I get is, "YOU'RE ONE OF THOSE SACD PEOPLE!" ... I didn't draw this line in the sand. I was pushed onto one side of it.
Don't laugh. I think that Hybrid DVD-A/SACDs could be made. I've looked at the UDF directory structure. It isn't impossible.
Mr Stefan Schreiber has outlined his ideas in European audiophile magazines and says he simply wants to end the format war to proliferate hi-rez music. Professionals have rejected his proposals of two-sided bonded discs, and some have even ridiculed his arguments. One of his faults was to suggest that anyone who preferred one format over the other should simply label the "wrong" format side of the disc...A double DSD/MLP mix or re-mix for every new project... I think some DVD-A's would be delayed for light-years then...
DVD-A players should ignore anything extraneous (SACD entries) in the directory tree, but I don't know if that's true with SACD players. (though, it should... Again, they're both UDF.)Just something like this:
Root:\2C_AUDIO
Root:\MC_AUDIO
Root:\AUDIO_TS
Root:\VIDEO_TS2C_AUDIO\2C_AREA1.TOC
2C_AUDIO\2C_AREA2.TOC
2C_AUDIO\2C_TAREA.2CH
2C_AUDIO\TRACK001.2CH
2C_AUDIO\TRACK002.2CH
2C_AUDIO\TRACK003.2CH
2C_AUDIO\TRACK004.2CH
MC_AUDIO\MC_AREA1.TOC
MC_AUDIO\MC_AREA2.TOC
MC_AUDIO\MC_TAREA.MCH
MC_AUDIO\TRACK001.MCH
MC_AUDIO\TRACK002.MCH
MC_AUDIO\TRACK003.MCH
MC_AUDIO\TRACK004.MCH
AUDIO_TS\ATS_01_0.BUP
AUDIO_TS\ATS_01_0.IFO
AUDIO_TS\ATS_01_1.AOB
AUDIO_TS\ATS_01_2.AOB
AUDIO_TS\ATS_01_3.AOB
AUDIO_TS\ATS_01_4.AOB
AUDIO_TS\AUDIO_PP.IFO
AUDIO_TS\AUDIO_SV.IFO
AUDIO_TS\AUDIO_SV.BUP
AUDIO_TS\AUDIO_SV.VOB
AUDIO_TS\AUDIO_TS.BUP
AUDIO_TS\AUDIO_TS.IFO
AUDIO_TS\AUDIO_TS.VOB
AUDIO_TS\DVDAUDIO.BUP
AUDIO_TS\DVDAUDIO.MKB
VIDEO_TS\VTS_01_0.BUP
VIDEO_TS\VTS_01_0.IFO
VIDEO_TS\VTS_01_1.VOB
VIDEO_TS\VIDEO_TS.BUP
VIDEO_TS\VIDEO_TS.IFO...Does anyone think that'd work?
...except for the fact that I might have just mocked up a 15GB disc... But... theoretically?
It seems that both DVD-Audio and SACD players expect incompatible lead-ins on layer 0 - so even if some players (Pioneer, Marantz, Onkyo, recent Sony's) can handle this hybrid, all the others will fail. My Panasonic machine DVD-RA71 (RP-91 U.S.) gets stuck in the CD layer of the Sonopress DVD-A/CD hybrid, the Denon A-1 cannot decide what to do and plays nothing... A DVD-A/SACD hybrid will yield the same results, I'm afraid...
There is room in the DVD Audio spec for dsd tracks.However the sacd security with the pit width processing is a substantial hurdle.
We all have to buy a new player for single sided 'Super DVD Audio Disc' replay.
is that a separate mechanism? Universal players use the same transport.... That's a point though: how would an existing universal player sort out the disc, even if it did work.
Is that the PSP results in an AM modulated signal on top of the normal disc readout signal. Both can be easily detected and filtered.
The PSP signal would have a lower carrier frequency than the normal signal derived from the disc.
Some of the security data is modulated into the PSP signal.PSP can also be used to create a visible image on the disc.
Because the PSP signal has a lower frequency it also means that the servo focussing mechanism must cope with the lower frequency PSP signal (wich affects the reflected light level).
The focussing servo loop mechanism must be less thight and must act slower to cope with the wider bandwith of the signal coming from the light detectors.
This means that the focussing (an also tracking) is less optimized for the datastream that is holding the actual information.
I suspect that disc and player tolerances must be kept thight to ensure stable operation. Wear and tear and long term in drift electrical circuits parameters can affect the playback reliability.
The many reported problems with TOC read difficulties and failures to recognize sacd layers could be related to the problem.
Welcome to the other side of the fence Michi.On the Hi-Rez Highway, it's the other way around.
Life's tough sometimes, isn't it?
You throw around cheap niceties like "regards" when you don't mean them.What's the other way around, huh? What the HELL is the 'OTHER WAY AROUND' here? What is so "reversed" here where my posting of information is patently FALSE and INAPPROPRIATE?
Yeah, that's right. I'm in your territory.
And PLEASE throw the 'REGARDS' out the window. If we were at a convention, and some DVD-A Jihad zealot shot me in the head, you'd go out for drinks afterwards.
This is a war in which the goal is to cause as much harm to the other side as possible, is it not?
Look Michi, you really need to get that chip off of your shoulder and lighten up.On the Hi-Rez Highway, SACD/DSD is praised to high heaven as the greatest thing ever while DVD-A is often described in terms you used earlier about SACD.
If you step back and think objectively about my message you'd immediately grasp the point.
As far as the "Regards" goes, it's called a closing/signature.
Anyone who has read anything I've posted should know that.And I have to say again, I was quite vocal in my incredulity about the supposed "MLP Flaw".
If I was a band-wagoneer, I'd jump on the "MLP Flaw" just because it was Anti DVD-A.
I wouldn't have spent my damn money on a damned 47Ai and 45A if it was dogma for me.
need to be repeated endlessly. Almost everyday, someone makes an inquiry that has been dealt with ad nauseum a week, a month or a year ago. There are also many new members. Just relax. I find DVD-A posters much more reasonable. Many of us hear listen to SACD and DVD-A. That is in contrast to Jazz IM who has no use for DVD-A as he sees only one "survivor" in the format "wars". Unforutunately, he is more representative of many on the hi-rez highway ie. Rich, Teresa(I think she is actually on the vintage forum now, as per Rich) even the moderator of this forum Chris, who clearly chooses sides, even as his SACD player is one of the many being repaired. If we want to giggle over the foibles of SACD, we should be able to, heaven knows many SACD supporters lord it over DVD-A.
First of all you should know that I think Rich is slime. And I think most people with some meat between their ears realize he's a moron, and I'm glad he's gone.Teresa, well, er, I haven't said this yet but she's a little bit nuts. So much so, that I'm somewhat afraid to say so. Her rantings about how anything in DSD is (was) orgasmically an epiphany, and that PCM causes children to murder their parents has given me the raised eyebrow many a time...
It's funny how she did a 180 to vinyl on that, though. She'll probably spend the rest of her days spinning about saying the word 'Vinyl!' over and over again... I don't know about her. That's why they call this the Asylum.
after I posted I realized I was a little late to the thread. For some reason the posts all of a sudden appeared even after I had been surfing and posting this morning. Anyway, glad to see things have worked out and settled down. Do I dare say I hope you and Patrick are happy!:') Sorry, couldn't resist.
heh
24 bits 96kHz capability.
SSL is still fuzzy!
Very expensive and still fuzzy sound.
32 bit floating point PCM DSP's with good algorithm's can give very good results. After the math is applied the signals much be rounded back into 24 bits and dither is used to linearize the result.
The cost for this is 2 bits from the 24bits available.
The signal degradation is insignificant if all is done right.There is no such thing as native DSD DSP's.
After a processing step DSD always ends up as pcm. To get back to a 1 bit stream dithering is needed wich needs at least needs a 3 bit PCM sample otherwise the signal will be compromized to much.
Too much processing steps with dsd and signal degradation becomes noticeable very quickly. This includes ALL processing steps in the complete chain. The ill effects of dsd editing during mastering and the steps applied in the player, for BM filtering or even level adjustments in the digital domain, are cumulative.Another problem is signal headroom. If two or more signals are added the dsd signal can saturate. They strongly advise to keep the signal level below -6dB of the full input range to prevent trouble.
PCM processing doesn't have this problem, a 24 bit signal handled by a 32 bit processor has plenty of processing offset.There are solutions for simple crossfades in dsd 'dsp' implemented.
The output is switched between the native dsd stream and the processed stream during the crossfade and if that's finished the signal is switched to the other native dsd stream. During this switching offset clicks at -50dB signal level are produced.
According two the philips engineer who presented this solution these clicks are insignificant.Frank
Another problem is signal headroom. If two or more signals are added the dsd signal can saturate. They strongly advise to keep the signal level below -6dB of the full input range to prevent trouble.
PCM processing doesn't have this problem, a 24 bit signal handled by a 32 bit processor has plenty of processing offset.
I fail to see how this is a problem for DSD but not PCM. The fact that a 32 bit DSP has 32 bits doesn't make any difference if the 24 bit source is fed into the 24 most significant bits of the DSP.
But anyway, I think an important point about bass management is getting lost in this discussion. Even if you use a 128 bit DSP running at 384 KS/s, you're still not going to get transparent bass management. The filter characteristics aren't dependent on the sample size or rate. Adding additional filters into the playback chain is always going to degrade the sound somewhat no matter how they're implemented.
Add two 24 bits pcm words and you could end up with a 25 bits word if the result overflows 24 bits. With 32 bit dsp that's not a problem. You need a lot of processing steps to run out of bits.
Feeding the 24 bits into the most significant bits of the 32 bit processor would be stupid.The end result has to be dithered down into 24 bits only once. Here you lose some of the bits and the s/n ratio degrades a little. But with 22 bits left (2 are used by the dithering) it's still very transparant and very linear.
The actual algoritmths implemented in dsp are very important to.
Bad filter design in dsp results in sonic degradation.With DSD it gets tricky. There really is little headroom for arithmatic.
In practise you end op with requantizisation or dithering down into 1 bit on many occassions after a processing step. And each time the signal deteriorates.The moment all the bits in the dsd stream are used up to express a high signal level less or no bits are left to express the finer details of the signal.
"Even if you use a 128 bit DSP running at 384 KS/s, you're still not going to get transparent bass management"
In theory not because there is still a finite resolution.
But in practise it will be very transparant if correct filter arithmetic is applied.
128 bit is a LOT of resolution and dynamic range.
A 32 bit dynamic range is already deadly.Frank
Add two 24 bits pcm words and you could end up with a 25 bits word if the result overflows 24 bits. With 32 bit dsp that's not a problem. You need a lot of processing steps to run out of bits.
Feeding the 24 bits into the most significant bits of the 32 bit processor would be stupid.
Frank, if you use the extra 8 bits as MSBs as you seem to suggest, you haven't eliminated any issues stemming from a lack of headroom on the input. You've only temporarily avoided the issues until you have to fit the result back into 24 bits, at which point you will either need to attentuate or compress the result. Further, without any additional LSBs, any processing steps that involve roundoff error will add to the noise floor. That defeats the whole purpose of using 32 bits. If the input & end result are 24 bit, then the advantage of doing intermediate processing at 32 bits is specifically to make sure that any noise and distortion introduced during processing is well below the 24 bit noise floor.
I believe that if you ask around, you'll find that it's industry standard practice to leave 2-3 bits of headroom in the final result during PCM mastering and 6 dB of headroom in the result of DSD mastering. I believe you'll also find that when higher bit depths are used during intermediate processing, the extra bits are usually used on the LSB end.
With DSD it gets tricky. There really is little headroom for arithmatic. In practise you end op with requantizisation or dithering down into 1 bit on many occassions after a processing step. And each time the signal deteriorates.
The moment all the bits in the dsd stream are used up to express a high signal level less or no bits are left to express the finer details of the signal.
I don't want to argue this too much since I really have no idea what Sony is doing in their DSD DSP. I would only point out that Sony recommends leaving 6 dB of headroom in the master specifically to avoid this sort of problem in the playback system.
I said:
"Even if you use a 128 bit DSP running at 384 KS/s, you're still not going to get transparent bass management"
To which you replied:
In theory not because there is still a finite resolution.
But in practise it will be very transparant if correct filter arithmetic is applied.
128 bit is a LOT of resolution and dynamic range.
A 32 bit dynamic range is already deadly.
Sure, 32 bits is more than enough given that the end result has to fit back into 24. But my point is that the desired filter characteristics don't depend on how precise the arithmetic is. No matter how many bits you use to represent a sample, the filter designer faces the same tradeoffs between the magnitude response, phase response, and other factors. Implementing the filter in 32 bit arithmetic instead of 24 bit arithmetic only means that the noise you introduce through roundoff error is much lower. It doesn't change the filter's frequency response, and thus it doesn't eliminate any audible artifacts of that response.
Dave
"Frank, if you use the extra 8 bits as MSBs as you seem to suggest, you haven't eliminated any issues stemming from a lack of headroom on the input."I assume that the recorded material uses the dynamic range to it's fullest potential in the max department but, of course, just under clipping.
Utilizing more defensive methods during recording to prevent problems in the rest of your production chain is fine. But they should consider upgrading to better processing equipment, like 32 bits fp processing."You've only temporarily avoided the issues until you have to fit the result back into 24 bits, at which point you will either need to attentuate or compress the result. Further, without any additional LSBs, any processing steps that involve roundoff error will add to the noise floor."
That's actuallly the case, but it's the best you can do to fit back the resulting data into 24 bit's with minimal signal degradation.
"That defeats the whole purpose of using 32 bits."
Not at all, the 32 bits processing is providing the headroom to get the maximum result from the recording.
It also obseletes the 'old school' thinking of the defensive recording practises that where neccesairy to circumvent the 'processing' bottleneck (be it analog or digital).
"If the input & end result are 24 bit, then the advantage of doing intermediate processing at 32 bits is specifically to make sure that any noise and distortion introduced during processing is well below the 24 bit noise floor."Exactly.
"I believe that if you ask around, you'll find that it's industry standard practice to leave 2-3 bits of headroom in the final result during PCM mastering and 6 dB of headroom in the result of DSD mastering."
"I believe you'll also find that when higher bit depths are used during intermediate processing, the extra bits are usually used on the LSB end."
I don't think that's true for processing.
For sample rate conversions,attenuation and filtering processes it's best to use the extra bits as lsb.
But for mixing (summing) and gain applications the extra bits are best used to maximize the headroom.
In practise the extra bits should be split equal at both ends of the pcm word.
In the last stage the file could be adjusted to use the maximum bitwidth by a fairly 'simple' bitshift before dithering it down to 24 bit.
This will always ensure maximum performance from the dac utilized in the players.
Frank
My Denon AVR-4800 uses an analog filter on an "analog direct" input stream, processes the bass only in the digital domain and then mixes in analog. The only effect I hear when it is turned on or off is more or less bass. Similarly, the analog "bass management" performed by the Paradigm X-30 active crossover which comes with my Servo 15 subwoofer comes at the cost of a subtle brightness which is just barely detectable.
Jim,I didn't mean to suggest that all filters are highly audible, only that implementing them in the digital domain doesn't eliminate the basic tradeoffs associated with finding the most transparent filter response.
Also, I've found that for me, the only kind of bass management I can live with is to run all of my speakers full range and use a sub to fill in the bottom end of their response. This seems to work best with subs that are designed to be run this way, as they tend to have a gentler LPF which blends better with the rolloff of the mains. For a long time, I bought into the theory that you need to HPF the mains because sound quality would somehow suffer if they were driven into LF excursions below their cutoff. But now I find that any kind of traditional bass management involving an LPF on the sub and a HPF on the mains (typically with fairly steep matched rolloffs) compromises low frequency imaging somewhat and makes the sub stand out more rather than blending in transparently. Obviously, this is less of an issue the lower your mains go, and yours go lower than mine, so your mileage may vary.
Dave
I'm not sure you should only focus on the 1 bit issue, there is a lot of interesting ideas for sample rate conversion. Several posts (by Graemme, Stephen) were made recently on the HRH which I really found useful. I had not realized that the sampling rate of DSD provides the best conversion path from and to any sampling rate of the 44.1 family, and probably the best conversion from 44.1 or 88.2 to 48 and 96k, using only integer multiples. If it wasn't for technical limits (upsampling to 14.1MHz is not easy, obviously), I think this methodology used in DSD could apply for PCM.Anyway, I'm sure you already know that stuff :)
Best
That's common knowledge. Oversampling for 44.1 at 256 times was done 10..15 years ago in 1 bit dacs with noise shaping.The stupid part of dsd is that it cannot handle 48k and it's multiples.
And these are becoming the standard bitrates for the future.
Frank, I'm not discussing the 1 bit aspect, nor the noise shaping (I'm not qualified), just the "translation" path.Correct me if I'm wrong:
Couldn't the DSD conversion tables work both ways ? (in theory at least) So if this is correct it should be able to handle 48k, and 96k "naturally", only through integer multiples. If I understood the information posted on the HRH:
48 x 294 =14112000 / 5 = 2822400
2822400 / 64 = 44.1Whether this can be implemented in practice, I have no idea. I just wanted to say that this is also a conversion path between the two PCM families, and probably better than going through interpolation.
If you maintain a 24 or 32 bit depth, and if you have to store any information at those crazy rates during the intermediate steps of your manipulations, however, you probably need a server and a rack of 120Gb hard discs..
With the advent of peer-to-peer and grid computing, this kind of processing is just around the corner.
Just musing...
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Eric
I think it is safe to say that Pioneer found that conversion of DSD to 88.2/32 PCM and leaving the signal in PCM was more transparent and more convenient than Sony's 1 bit method. This is clearly a lossy conversion, as is any conversion of DSD to PCM other than 1 bit. But, if you do a 1 bit conversion you run into the headroom issue. Conversions from PCM to DSD, on the other hand, should be lossless provided that they are feasible.
as you know I don't trust Pioneer all that much with their implementation of standards :)
But it makes sense, because DSD is apparently compatible with all variants of the PCM 44.1 family.Best
Eric,It's only integer multiples of the sampling frequency, it's decidedly not integer arithmetic for the actual data reduction.
As a side effect of the DSD--> PCM translation you are left with the sonic signature of DSD (diminishing SNR as a function of frequency) on the resultant PCM output since PCM has a linear noise response inside the pass band.
No one has yet demonstrated that the process of Super Bit Mapping Direct gives the same result as a capture at the native rate.
Keep in mind, that to do an SBM Direct transfer, you have to do a 5x oversample -- it cannot be done natively. In other words, insufficient data exists in and of itself within the base sampling rate to reproduce a substantially lower fs.
Further, if you want to goto 192kHz you have to use 5x, followed by 2x oversampling.
In the end, you have a 147:24 reduction for 96K PCM (from 5x oversampling) and a 147:24 reduction for 192K PCM (from 10x oversampling).
Here I have assumed that sampling depth follows current practice, and results in a 24-bit PCM word.
Just my opinion!
I was not disussing the transfer from DSD to PCM, but only the methodology, which makes sense (to me). There is no reason you could not follow the same methodology, even retaining a 24 bit or 32 bit word. The conversion paths should work even within and between the two main PCM families (forget the Laserdisc for now :)Just a quick note on 192k: It would probably make more sense to oversample x2 the final 96 than go through 28MHz.
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And yet, the SBM direct process does indeed do a 5x, followed by a 2x oversample to generate enough data points to have a 192K datastream.There is (IMO) no point to 2x oversampling a 96K PCM datastream to get to the 192K datastream.
As far as a final value.... As I understand it, it's merely a summation of 1s (+2v) and 0s(-2v) divided by the number of samples.
nsamples == 147
1 samples == 100
0 samples == 47Aggregate voltage == 200 -94 or 106.
Divide aggregate voltage by number of samples (106 / 147) which is approximately .72 which gets encoded as a signed 22-bit value, then dithered out to 24 bits.
I know how the basics of the process works, I just don't think it's a sound idea above 48kHz fs. I can't prove it either ;-)
Regards,
JohnYour knowledge of DSD is far greater than mine.
The increased I/O processing added for that extra x2 step is enormous. There must be a good reason why there is an oversampling of x5 and x2 instead of x2 on the final 96k result, probably to prevent interpolation as much as possible.
The more I think about it, the more I find it intriguing.. 192k is the only rate where things do not translate smoothly with that method.As for the rest, I hope someone can challenge your post better than me. I'm out of my league at this point. : )
(Damn, where's Frank?)
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You don't get more information from dsd by oversampling it five times.
The oversampling is done to make things easier on the math applied.
Nope, I'm talking about having sufficient data to make the reduction meaningful.
"No one has yet demonstrated that the process of Super Bit Mapping Direct gives the same result as a capture at the native rate."I have listened for differences with DSOTM the 'redbook' layer of the hybrid against the 20th anniversary edition.
The latter has better detail and has a wider 'soundstage'.The cd layer of the hybrid has less depth and the instruments are sculpted a bit 'smaller'. It sounds a bit recessed.
I think the CD layer of the new DSOTM is the version of the "Shine On" box set. It would be interesting to compare the files using EAC.Best
Would make the file different anyway.The differences between the 20th anniversary and the redbook layer are very small.
I suspect the same source is used.
I will run some more comparisons at louder volume levels.
I used EAC "compare WAV" feature between the CD layer of some SACDs and the equivalent "CD-only" edition. For example I compared the redbook layer of the Ray Brown Monty Alexander etc from Telarc, with the CD (which also has a good bonus disc of Ray Brown). These two were published by Telarc, and were released almost simultaneously. So what you find is that all the tracks are identical... except for 638 "missing samples" in each track in the CD edition. These missing samples are located at the very begining of the track (usually in a silent place), you cannot hear anything in that section when you play it, and I cannot figure out what they are.At one point I thought of posting this on the HRH to ask what it could be, because I didn't feel like being accused of trolling.
Anyway, I'm afraid record companies won't bother re-mastering the CD layer so if you buy a hybrid SACD reedition, you only get the latest remastered CD version of that title if there was one.
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The 20th anniversary set and the new release are new masterings, and therfore no longer can be considered identical.
We'll never be able to hear much more than 20 bits.
I've heard Lexicon has got a Universal play out, or in the works? I guess it comes down to that video circuitry and whether, or not, a high end manufacturer focusing on SACD wants to go that route.
with channel trim only on the player and the speakers in an ITU configuration - maybe the Linn Unidisk. Or upgrade my receiver to allow two 5.1 inputs, keep my Denon DVM-4800 and go with the Krell if it does have analogue bass management and sounds good.
from channel level adjustments in sacd players if it's unknown how it is implemented.Frank
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