Topic: Korg Mr-1 vs Sony PCM D-50

[craig_c]

I found an MR-1 at a price I couldn't pass up, but I'm wondering if I should look for a PCM D-50 and unload the MR-1. I was hoping to record industrial sound and shift the pitch/tempo extremely downward, like make a 10 sec. sample be 30 secs, and approx. 3 octaves lower. I thought doing this in DSD would provide the least resample artifacts. Turns out I can't do this edit in DSD, I have to covert to PCM format and edit in my DAW.
SO, the question is this: will the PCM file from the Korg be of a higher quality than if I had recorded using a PCM recorder such, as the D-50, (assuming same sampling rates) and result in fewer artifacts?

Thanks for any advice.

Craig

[headroom]

DSD 1 bit is with a lot of noise shaping. And the PCM 50 D has excellent Amps Limiter and runs cool and long with battery. Great sound I think only Metric Halo ULN stuff is better

[craig_c]

"DSD 1 bit is with a lot of noise shaping."
Sorry, but this newbie doesn't know what "noise shaping" is. PCM doesn't use noise shaping? I read the Wikipedia entry on "noise shaping" and I gather that it is a method to increase the signal to noise ratio. I thought that DSD samples so fast that it is a more accurate representation of the sound, minimizing quantization errors better than PCM.
Are you saying that a good quality PCM recorder has better signal accuracy?

Craig

[headroom]

1 bit DSD Recording is very very agressive and i d not know your a newbie...
DSD is 2.8 MHZ and PCM is 44.1 /48 /96/192 kHz converting to lower samplerate will allways degrade the sound because of uneven Numbers. 88.2 to 44.1 Khz is not a problem because is exactly  1/2 rate.
I have 2 PCM 50 and the only drawback is the lack of Phantom Power 48 Volt for external mics.
And the 50 have the possiblity to play ist  slower or  speedup without pitch shifting

[craig_c]

No reason you could know I'm new at this. Digital imaging I understand, but I'm learning audio.
Could you expand on what you mean that DSD is aggressive?
Also, I want to radically shift the pitch. For instance, I want to record record a car engine and shift the pitch/tempo way slower. If I convert the DSD file to a 192 PCM and shift the sound downward in my DAW, will I get a better result than if I had recorded at 96KHz in PCM and then resample the file -- less artifacts?

[headroom]

No reason you could know I'm new at this. Digital imaging I understand, but I'm learning audio.
Could you expand on what you mean that DSD is aggressive?

To much very complicated technial stuff to explain.

Also, I want to radically shift the pitch. For instance, I want to record record a car engine and shift the pitch/tempo way slower. If I convert the DSD file to a 192 PCM and shift the sound downward in my DAW, will I get a better result than if I had recorded at 96KHz in PCM and then resample the file -- less artifacts?

Dont think so for car sound and stay in 24 bit until ther end process...

[headroom]

DSD provides an extreme example that shows how SNR is not determined by bit depth alone:

DSD is a 1 bit system with a sample rate of 2.8224 MHz. When measured over its entire bandwidth (1.4112 MHz) its SNR is less than 6 dB. However, DSD achieves an excellent SNR (up to about 120 dB) when measured from 20 Hz to 20 kHz. DSD uses very aggressive noise shaping to move the quantization noise out of band and into high frequencies.

The in-band SNR limits of a digital system are determined by the following:

1) Bit depth (about 6 dB per bit)
2) Sample rate (about 3 dB for every doubling of the sampling frequency if noise-shaping is not used)
3) Noise Shaping (substantial improvement - especially at high sample rates)

Noise shaping is only effective when there is spectrum space available to "hide" the noise. The noise shaping that is often used on 44.1 kHz CDs moves much of the quantization noise the the band between 20 kHz and 22 kHz. Packing the noise into this 2 kHz band can substantially improve the perceived SNR as well as the measured 20 Hz to 20 kHz SNR. 96 kHz has a much wider band available for noise shaping - (20 kHz to 48 kHz). 96 kHz has a 26 kHz band available for noise shaping while 44.1 kHz has only 2 kHz available. DSD has 1391 kHz available for "hiding" quantization noise.

Noise shaping must be re-applied whenever a signal is modified (gain change, EQ, etc.). Cascaded noise shaping processes can be very damaging due to rapid build-up of out-of-band noise.

For this reason DSD in not a good format for use in recording and editing.

Noise shaping is rarely used at 96 kHz because most 96 kHz systems support 24-bit data. For this reason 96 kHz 24-bit systems have gained widespread acceptance in pro-audio applications. 96 kHz 24-bit audio can be edited easily without needing to apply potentially-damaging noise shaping.

http://www.hydrogenaudio.org/forums/index.php?showtopic=75887&mode=threaded&pid=665724

[craig_c]

Well, I'll have to dig in and study this. When you say that DSD is not good for recording, I'm confused that so many people in reviews seem to think that their DSD recordings sound much better than PCM recordings. Is this due to a sort of audio placebo effect — that since they bought this gear they are expecting it to sound better?

And what happens during editing to the displaced noise that is shifted to higher frequencies?

Also, thank you very much for your detailed reply.

Craig

[headroom]

All digital systems have a different sound even with the same processors but you need a trained ear and very high resolution chain to hear this.
For a hobbist its not necerlery to go into the Highend Stuff wich is very expensive. So what is your goal? Just strart recording sound make some expiriences, but do not go to deep into all this technical blabla. We are living in times of the digital proletariat. Example Poeple with a Pro camera think they are professional photographer but dont know ho to viewing like a Pro, wich is a long process.

Only a good recorder makes no a difference.
It`s the sum of everthing  its the Power Source the cables and the Mics there placement, the Micamp and the recording technique skills and a very well trained ear wich makes the difference. There are so many mastubators in  all forums wich not hear the difference from cold amp and a heated up amp. For maybe 2 Minutes the sound is bad going up and after 15 Minutes it is dropping down. The SONY PCM 50 not, it uses just very little power current aka no heat.

[Nick's Picks]

I have run an mr1 on and off again for the last few years.  I find it a very, very good sounding deck.  I would master in DSD and bring it straight to redbook via the software if I didn't need to do any tweaking (like adjusting levels or whatever).
the results spoke for themselves to my ears

[headroom]

What is to Author with SACD?  You send the tracks to Sony, who charge you a small fortune, and they do the pressing - you cannot now, or will ever be able to - create your own.
No Burners
No Drives that will play the DSD streams
No software players.

Why not stay all the time in HIREZ  96/24bit and play it it with DVD-A or over Firewire ? Downsampling DSD to Rebook will shure not sound equal like the Original DSD stream.

[Shadow_7]

I have the MR-1000.  I didn't really like the MR-1 version based on specs.  Power issue(odd battery), no preamps, 3.5mm input, and no onboard mics?  And not much different in price (at that time) from the MR-1000.  Plus I got a good deal on a used MR-1000.

DSD is nice, the book says a larger dynamic range than 24 bit.  Editing DSD is for the most part not an option, so you have to convert it to PCM anyway.  Although the most recent version of audiogate has some extra editing like ablities that can be applied to DSD / SACD outputs.  And a few editors are becoming DSD capable.  But for me I convert to 24 bit @ 96 kHz to edit. 

The difference to my ears is the transient detail.  There just appears to be more of it, even on relatively low sensitivity mics.  And even after conversion / downsampling.  Until I double my computing power (3yo system), 24@96 is just easier / quicker to do.  And something that you could pass off to someone else unedited and they can do something with it themselves.  But if you don't record in DSD and convert there's really not much difference between it and other options, outside of price.

[page]

1 bit DSD Recording is very very agressive and i d not know your a newbie...
DSD is 2.8 MHZ and PCM is 44.1 /48 /96/192 kHz converting to lower samplerate will allways degrade the sound because of uneven Numbers. 88.2 to 44.1 Khz is not a problem because is exactly  1/2 rate.

I think your committing the general fallacy of logic with PCM... It's true you are getting more samples per second, but your not sampling each frequency more often per second. Degradation is largely based on your sample rate converter routine much more so then what parent frequency you recorded at.

A better reason to choose between 88.2khz and 96khz would be what you're trying to record in the missing band (such as special nature recording situations and sound effects that will be slowed down later in post processing) or the final application (e.g. DVD Video which doesn't support 88.2).

[ghellquist]

... converting to lower samplerate will allways degrade the sound because of uneven Numbers. 88.2 to 44.1 Khz is not a problem because is exactly  1/2 rate.

They say a little knowledge is more dangerous than no knowledge. In this case you are totally wrong today.

There is no advantage in sound quality in keeping any special ratios when converting. There is an old myth that even ratios ( 44.1 to 88.2 has the ratio 2 ) would sound better. This is only a myth today. It was partly right in older times when computers used to bee much slower than today.

It might be true that you could, possibly, save a little computer processing time when doing even ratios. I would not bet on it though, not having written any commercial quality sample rate converting software. I bet our friend headroom has never written any sample rate converting software either. Computer processing is plenty available today, so that is not an issue in any normal processing.

// gunnar

[ghellquist]

1 bit DSD Recording is very very agressive ... 

Again, a little knowledge may be very dangerous.

DSD recording is not aggressive. Not in the way a lion or a dog could be. And it does not aggressively modify any sound. It is simply a technique to catch and transfer sound samples. It has advantages and disadvantages, like any other technologies.

And like any other use in a technical application, it is generally not the theoretical limits that sets the quality borders but the practical application and most often the design tradeoffs in actual appliances.

I believe there is a common understanding among recording engineers that the differences between DSD and PCM are small. Not insignifacant in every case but still small. Especially if you compare to the common consumer formats, say mp3.

// Gunnar

[ghellquist]

Back to the original question.

SO, the question is this: will the PCM file from the Korg be of a higher quality than if I had recorded using a PCM recorder such,

No.
The quality diffences will not to any large amount be dependant on if you record PCM or or DSD and then transfer.

The limiting factor in both these cases is the recording box. All boxes you can buy has been built to a set of limiting factors. Most often the factors are things like retail price and size.  You can get better recordings by getting a more expensive, larger box, say a Nagra VI or one of the Sound Devices boxes or if you preferr that path the MR1000.

The MR1 as such is a small box, at a low price using little energy. This will limit the sound quality. The Sony is similar in price, size and energy usage and as such basically falls in the same category. For some usages either box may be the choice.

Now again, is this going to be your most pressing problem. My guess is not. I thing you will be able to get good sounds from that box. But there really is only one way to know, test for yourself.

Personally, I do not find the hassle with DSD worth it. I go with 24 bit 44.1 kHz and is happy with that using a SD722. My choice.

// Gunnar

[Shadow_7]

Personally, I do not find the hassle with DSD worth it. I go with 24 bit 44.1 kHz and is happy with that using a SD722. My choice.

The only hassle of sorts is that you have to preconvert before editing.  Which can take as long as your recording depending on your computing horse power.  With the MR-1000 you can always tether another device on the output if you need immediate lower resolution results for quicker turnaround.  Or just record in the lower rates, which sort of defeats having DSD and makes most options roughly equal, sometimes better.  And as the software catches up most of the hassle is moot.  Storage is cheap now.  Computers are fast enough to handle DSD audio in realtime.  There's even a few laptops with DSD soundcards now.  Audiogate can play DSD audio while downsampling to 24@96 in realtime on my 1.9GHz dual core desktop through an M-Audio Delta 44.  The only real hassle is that there's no current FREE / OSS options to convert / edit DSD.  But audiogate comes with the device and does the job, even if you do have to boot windows to run it.

[headroom]

Noiseshaping sould be used in the very last stage of the signal processing and not heavily in advance, like in the DSD process.  If you equaize. normalize etc. it can have more and more artifacts as you work on the files. Als with DSD you can record only 2 channels and not mix donw serverals. Bits and bits of music are not just only numbers on and off.
Solid state players sound better then the CD ripped with EA.
Sample rate converting can be done DA/AD or with software. Its a very critical process where a lot of damage can occur. Check here for different plugins:
http://src.infinitewave.ca/

[ghellquist]

The only hassle of sorts is that you have to preconvert before editing.

Sorry. I should have written differently. I should have said that my ears are happy with 24/44.1 . Period.
// Gunnar

[ghellquist]

Noiseshaping sould be used in the very last stage of the signal processing and not heavily in advance, like in the DSD process. 
...

Sorry headroom. Could you step back a little. Theory is nice in many ways, but come back when you have used a DSD recorder in earnest. Things do not behave exactly as a non-complete theory predicts.

Als with DSD you can record only 2 channels and not mix donw serverals.

Sorry, you are off again.
http://www.digitalaudio.dk/ax24.htm
http://www.merging.com/products/show?product=1

// gunnar

[headroom]

Ok I see some live on Planet DSD if you have the cash...

[Shadow_7]

The only hassle of sorts is that you have to preconvert before editing.

Sorry. I should have written differently. I should have said that my ears are happy with 24/44.1 . Period.
// Gunnar

I'm an old trombonist that's played for decades in acoustic groups, I've only recently got into the recording thing.  And to me a recording still sounds like a recording, anything 24/96 and down IMO.  DSD less so, but still not perfect.  Better reproduction gear helps, but it's still got that something is missing recording flavor to it. 

There's a lot of really good gear out there.  And there's a number of noisy tracks with phones ringing, sirens in the background and other things that has me turning off the radio and checking the phone just to verify from wence the sound is coming from.  But still not perfect for sounds that are less synthesized and are familiar to me.  Don't get me wrong, 24/44.1 is good.  And good enough for a lot of purposes.  But I can still tell the difference between 24/44.1 and 24/96 and DSD.  When the content is the same and on good reproduction equipment.  Not to mention that a DSD capable machine makes a really great MP3 player.

Planet DSD is not that expensive IMO.  It's not cheap.  But most of it can be had for < $1K.  Comes out to less than $500 a channel.  Lots of mic+preamp options that are much more expensive, for far less facility.

[craig_c]

Thanks for all the great input and discussion.
As for the expense of DSD land, I bought my MR-1 used for $150. Pretty damn good. True, I can't edit in DSD, but I can export to 24-bit/192 kHz for editing.
I don't plan to record concerts, symphonies, jam sessions, barking beetles, or 9 hour chanting sessions. I'm not trying to burn DSD discs.
I will do short samples of industrial sound sources, e.g. car engines, electrical motors, hot water heaters, water in pipes, gas cylinders struck with a mallet, etc. etc.  I will then shift the pitch and tempo radically in my DAW (LiveLite), and save as samples to use in musical compositions: engine noise as percussion, gas cylinder as harmonic tones. So my primary concern is to minimize the resampling overtones and artifacts when radically resampling the recordings.

Would love to have more input on this.

Thanks,
Craig

[taperj]

No Burners
No Drives that will play the DSD streams
No software players.

Pure misinformation there. ALL Sony Vaio laptops can render DSD disks natively as can PS3. You don't have to burn SACD as you can just burn a DSD disk, in addition to audiogate there are other free tools to do this. When I got my mr-1 I successfuly burned and played many DSD disks.

http://en.wikipedia.org/wiki/Direct_Stream_Digital#DSD_Disc_Format

"Transferring this signal to a recordable DVD disc with the appropriate tools, such as the Korg MR-1/1000/2000-bundled 'AudioGate'-software, will render a DSD Disc. Such discs can be played back/transcoded to PCM on the fly on most PCs (this requires a DSD plug-in for Windows Media Player) and in native DSD on Sony VAIO laptops and PlayStation 3. Moreover, Sony has released a new SACD-player (autumn 2008), the SCD-XA5400ES, which fully supports the DSD-disc format. "

As far as editing DSD in a DAW goes, I haven't tried it but my Cakewalk Sonar 7 plainly states it has a sony DSD import plugin.

[Ozpeter]

Before becoming too concerned about what 'better-than-CD' formats bring to the audible table, http://drewdaniels.com/audible.pdf is worth a read.

[craig_c]

While I understand some of the issues in a discussion about qualitative differences in formats "beyond" Red Book Audio. I'm hoping someone will chime in on my main issue which is how to best minimize resampling artifacts when radically shifting pitch/tempo.
Imagine using actual tape. I want to do the equivalent of recording at normal speed, and playing back at half speed, or 1qtr speed, slowing the audio, shifting pitch lower.
Now I want to do this digitally with minimum "wow", reverby type extraneous sounds.

Help, anyone, anyone?

thanks,
Craig

[wagnerian]

I'm hoping someone will chime in on my main issue which is how to best minimize resampling artifacts when radically shifting pitch/tempo.

I am in no way a technical guru, but what I understand is that DSD is artifact-vulnerable format to edit. So just record in PCM, or keep in mind when you record in DSD that you must convert the recordings into PCM before editing.

Now, the real matter for you is what kind of PCM resampling algorithms and softwares are best for the purpose. Sorry that I know little about that.

[craig_c]

I know I can't edit in DSD.
Hmm, my original question is about will I get a better result recording with the Korg, converting the DSD to PCM 192 and editing in my DAW, or recording in PCM at the highest available rate in a Sony PCM D-50. Either recording will be severely resampled. So is it better to do the original recording in DSD, or does it really make no difference, in which case it might be better for me to unload the Korg and purchase the D-50 for its other advantages in field recording.

Thanks,
Craig

[page]

I know I can't edit in DSD.
Hmm, my original question is about will I get a better result recording with the Korg, converting the DSD to PCM 192 and editing in my DAW, or recording in PCM at the highest available rate in a Sony PCM D-50. Either recording will be severely resampled. So is it better to do the original recording in DSD, or does it really make no difference, in which case it might be better for me to unload the Korg and purchase the D-50 for its other advantages in field recording.

Thanks,
Craig

While I'm not sure I can really address your question, I'd keep the korg and record in PCM with it (it's an option, I've tried it on mine). Then the artifacts induced during conversion will be the same (since it's PCM either way). A difference between the two would be how well it's A/D functions or any noise in the analog stages, but thats outside of your question.

I'd keep the korg since you have it, and you won't have to spend an additional $200-300 to get the D50. Now, if you try recording in LPCM with the korg and don't like the results of either of the two areas I mentioned, then it would be time to look at the D50.

[craig_c]

I wonder if the PCM file created by AudiGate from the DSD file is better than one recorded by either the Korg or the Sony?
Any ideas?

Craig

[ghellquist]

Craig,
   you have a valid question but you consistently get answers on other things in this thread. Part of it is because no person answering you has done what you are trying to do, and simply does not have a clue. This includes me. We are all trying to inferr from other knowledge and experience, and not beeing very methodical in doing it. I will give it a try, for what little it is worth.

First you need to record things. You will get a really large quality difference depending on what equipment you use and how you use it. Mics, mic placement, mic preamp, recorder all are aspects to consider. But we cannot really know how it will affect your situation. I know a lot about the things I record ( symphony orchestras ) but I never recorded special effects.

Secondly you are planning to do some heavy handed modification of the sound. In a way, what  you are looking for is not high-fidelity reproduction, instead you are looking for the artifacts of the change. Very different from what most of us do. No question, it is perfectly valid, only different. I know there are different programs out there that can slow down your sound using quite different algorithms. My personal guess is that the choice of program and how you use it is the single most important discussion in this chain of processing. Compared to that PCM or DSD or whatever is extremely unimportant.

So what you need is to test with actual material and probably in different programs. My guess is that you will be best served with programs going a bit outside the normal envelope. This suggest to med programs like Ableton Live. Or perhaps one of the more specialized programs used in working on creating sample libraries, of these I know nothing.

So, in short. Just do it. Check the results. Modify if you need to otherwise not. And why not come back here and tell us the results.

// Gunnar

[craig_c]

Gunnar, thanks for the very frank and on-the-point reply.
I was coming to the same conclusion as you described, i.e. I need to do the tests. Clearly the severe tweaking I intend to do will depend on the methods I employ, especially the sophistication of the algorithms of the DAW.

I guess I was hoping someone could speculate, based on better theoretical knowledge than I have, about the relative quality of recording at 192KHz in PCM in a device like the Sony, vs recording in DSD and converting to PCM. My assumption is that the cleaner PCM file I have to start with will yield fewer artifacts in my resampling.

I'm starting to worry that using DSD as a source will yield a PCM file that has so much noise moved into the upper frequencies, that when I downsample the file that noise will create more artifacts than if I had simply started with a high quality PCM file. Since I don't own another recording device, I really can't test this.

I do have a Sony Pro Walkman, so possibly I could record on that for comparison (but perhaps with unfortunate tape hiss).

As for mic placement, for a lot of what I will record I will be using piezo contact pickups.

This has been a very interesting discussion of ideas.
If anyone else wants to chime in, I still appreciate the input.

thanks,
Craig

[Shadow_7]

In terms of effects.  Sox has three related to what you sound like you want to do.

pitch - changes the pitch, but keeps the same timeline.
tempo - changes the durations of things, but keeps the same content
speed - like a variable speed dial on a tape player.  Takes the timeline and either stretches or squishes it.  Changing both pitch and tempo in the process.

$ sox input.wav output.wav speed 1.00011

(what I need to do to bring my MR-1000 results to run about the same speed as my FH1 camcorder.  Less than 1 second per hour difference, but drummers at 60p and the drift is noticeable in under ten minutes)

As far as DSD, most DSD recorders can ALSO record the standard PCM files.  If you don't like the DSD option, you don't have to use it.  And if you're not paying extra for it, why not have it?  As far as editing and players, that'll improve in time.  Low resolution PCM files wont.

[Ozpeter]

how to best minimize resampling artifacts when radically shifting pitch/tempo.
Imagine using actual tape. I want to do the equivalent of recording at normal speed, and playing back at half speed, or 1qtr speed, slowing the audio, shifting pitch lower.

If you want to do exactly what tape slowing does, then there's no problem at all.  I'd suggest playing around with Reaper, which is a small download and a non-invasive installation but which has good tools for this kind of thing.  At its most basic, there's a slider to change the playback rate, and that should introduce no artifacts at all.  But it can also use the Elastique algos for stretching without pitch shifting or pitch shifting without stretching, and they have a good reputation (though any such operations are bound to produce less than perfect results).

[Nick's Picks]

What is to Author with SACD?  You send the tracks to Sony, who charge you a small fortune, and they do the pressing - you cannot now, or will ever be able to - create your own.
No Burners
No Drives that will play the DSD streams
No software players.

Why not stay all the time in HIREZ  96/24bit and play it it with DVD-A or over Firewire ? Downsampling DSD to Rebook will shure not sound equal like the Original DSD stream.

If you say so.
I couldn't hear any difference on my system (playing the raw files off the deck vs. the same dsd> 16/44.1 wav).  I"m just speaking for myself, my ears, my playback system which has been consistently trustworthy for its accuracy....according to me.  ;-)
as individuals, with our own sense of perspective.  I have no problem understanding if you heard these differences on your system when you tried it.   so.., bottom line.  Opinions can be given.  Truth is completely unobtainable.  A techie could show me the differences on paper, and while numbers dont lie I know for near certainty that I could not pick out A vs B blind.
The best you can hope for is a significant agreement from a lot of first hand users one way or another.

[craig_c]

RE: Sox, and Reaper. Are these apps available on the Mac side?
Are there similar?
I have Live Lite 7, Audacity, and Garage Band, but I'm really at the beginning stages of learning any of these.

Craig

[craig_c]

OK, so I took a moment and looked up the compatibility of the SoX and Reaper. I should have looked before posting my question. I'm not much of a command line guy but Reaper looks pretty compelling.
Thanks for the referrals.


Craig

[headroom]

What is to Author with SACD?  You send the tracks to Sony, who charge you a small fortune, and they do the pressing - you cannot now, or will ever be able to - create your own.
No Burners
No Drives that will play the DSD streams
No software players.

Why not stay all the time in HIREZ  96/24bit and play it it with DVD-A or over Firewire ? Downsampling DSD to Rebook will shure not sound equal like the Original DSD stream.

If you say so.
I couldn't hear any difference on my system (playing the raw files off the deck vs. the same dsd> 16/44.1 wav).  I"m just speaking for myself, my ears, my playback system which has been consistently trustworthy for its accuracy....according to me.  ;-)
as individuals, with our own sense of perspective.  I have no problem understanding if you heard these differences on your system when you tried it.   so.., bottom line.  Opinions can be given.  Truth is completely unobtainable.  A techie could show me the differences on paper, and while numbers dont lie I know for near certainty that I could not pick out A vs B blind.
The best you can hope for is a significant agreement from a lot of first hand users one way or another.

Ok I was not on the newest level with DSD mix playback possibilties thanks for correcting my fault.
I made was typing it wrong: DSD sound is agressive what i forget to type the rest of my meaning, it is the noise shaping algorythm that is agressive. Its very clear that HIREZ sound very analog like DSD or any other format.
I cannot belive you did not hear a difference between Original DSD and downsamplet to 16/44.1
(playing the raw files off the deck vs. the same dsd> 16/44.1 wav)
Did I not understand  english, my native language is Swiss. Please enlighten me....

[Nick's Picks]

What is to Author with SACD?  You send the tracks to Sony, who charge you a small fortune, and they do the pressing - you cannot now, or will ever be able to - create your own.
No Burners
No Drives that will play the DSD streams
No software players.

Why not stay all the time in HIREZ  96/24bit and play it it with DVD-A or over Firewire ? Downsampling DSD to Rebook will shure not sound equal like the Original DSD stream.

If you say so.
I couldn't hear any difference on my system (playing the raw files off the deck vs. the same dsd> 16/44.1 wav).  I"m just speaking for myself, my ears, my playback system which has been consistently trustworthy for its accuracy....according to me.  ;-)
as individuals, with our own sense of perspective.  I have no problem understanding if you heard these differences on your system when you tried it.   so.., bottom line.  Opinions can be given.  Truth is completely unobtainable.  A techie could show me the differences on paper, and while numbers dont lie I know for near certainty that I could not pick out A vs B blind.
The best you can hope for is a significant agreement from a lot of first hand users one way or another.

Ok I was not on the newest level with DSD mix playback possibilties thanks for correcting my fault.
I made was typing it wrong: DSD sound is agressive what i forget to type the rest of my meaning, it is the noise shaping algorythm that is agressive. Its very clear that HIREZ sound very analog like DSD or any other format.
I cannot belive you did not hear a difference between Original DSD and downsamplet to 16/44.1
(playing the raw files off the deck vs. the same dsd> 16/44.1 wav)
Did I not understand  english, my native language is Swiss. Please enlighten me....

Just being honest with myself.  I'd love to say "yea, I could hear a difference.  the raw DSD sounded way better".. , but I can't.  I didn't hear such a thing.  the DSD > 16bit file was indistinguishable to me.
Have you sat down w/a high res playback system and tried it blind ?

[Shadow_7]

There is a difference.  Unless you're playing everything back on a 16/48 soundcard.  Then of course they sound the same, they're all being downsampled to roughly the same rate.

I can tell a noticeable difference between DSD and 16/44.1.  It's not as dramatic as say 16/44.1 to 8/8(can you say CD vs. cell phone).  The 16/44.1 downsampled just sounds better to me than 16/44.1 that was recorded at and NOT downsampled.  Maybe not on the highest of high end gear, but most DSD options are still sort of mid-grade gear(relatively).  But then again it is very noticeable if you play those various versions back on the device that recorded the source material.  And on monitors/speakers that are mostly capable of reproducing it.  i.e. the grind grind grind of winding a disposable camera sounds like the source and not like someone with the walking farts near the mics.

[headroom]

Just being honest with myself.  I'd love to say "yea, I could hear a difference.  the raw DSD sounded way better".. , but I can't.  I didn't hear such a thing.  the DSD > 16bit file was indistinguishable to me.
Have you sat down w/a high res playback system and tried it blind ?
[/quote]

Maby the limited factor is in your recordings. We have done test with Linn stuff ( Linn 192k24 bit recordings) CD and Networkplayers with the 7k Networkplayer we were all  hearing that HIREZ is clearly better then downsamplet to Redbook. And the CD had no chance at all.

[Nick's Picks]

maybe

[headroom]

maybe

No it was a joke. Dont know maybe it is the setup. Can you Monitor with high resolution Headphones?
I use the Audio Technica ATH AD 900 they are stellar also the 700 is very very good.

[Nick's Picks]

i'm not the headphone type,. and dont have anything better than low end Senn and Beyers laying around.  I guess my best are my M-Audio in ear monitors.  its their lowest end, but still the best sounding I have. 
I prefer to sit in front of the stereo.

[headroom]

Sorry headroom. Could you step back a little. Theory is nice in many ways, but come back when you have used a DSD recorder in earnest. Things do not behave exactly as a non-complete theory predicts.

We dont listen only the Recordings, the real acid test is the Original Sound versus the recorded one.

AND:


The inherent flaw with the DSD amplitude quantization however,
is that it is much noisier than the quantization noise in a PCM system. This problem is addressed later though, with respect to each systems frequency bandwidth, dither and noise-shaping abilities all tools in diminishing the effects of quantization error noise to a practically inaudible level.

Whichever value the system assigns will obviously be a distortion of the signal, and the ensuing quantization error is audible as a rough, granular soundî (Harley 538). DSD unfortunately suffers from more of this effect than PCM; however, as seen later, is able to combat the issue much more effectively.

192 kHz PCM and DSD can displace the majority of this noise above the audible spectrum completely, reducing the audible effects by much more than the 3 dB that lesser quality PCM produces. As previously mentioned,
one major pitfall during DSD quantization is that ìthe quantization error and noise associated with sigma-delta modulation is very large, resulting in significant quantization noise, however, unlike the PCM structures
this ìnoise can be shaped such that virtually all noise power falls outside
the range 0-20 kHz, well above the range of human aural perception
(Nuijten 27). An even further distinction separating DSD and PCM is the ability of DSD filters to shape the dither with a much less complicated linear equalization curve  that is not forced to rely on less audible frequency regions below 20 kHz.


Since the sampling process is essentially a quantization of samples on a time scale rather than on an amplitude scale, signals occurring between those instances are cropped. In a 44.1 kHz system the time between each sample is fairly large, with approximately 22.68 μs between each sample. On the other hand, 192 kHz PCM samples are only 5.1 μs apart, and DSD has an even greater time domain resolution with a mere .357 μs between samples. As a result, the 192 kHz PCM and DSD systems are better suited to respond to transients accurately. How this response time actually affects the systems ability to reproduce a transient, as each system was fed a -6 dB block input (click) of a 3 µs duration. The resulting graph shows that the DSD and 192 kHz PCM systems respond the fastest and most accurately; whereas the 48 kHz PCM system not only distorts the signal, but also takes a much longer time to even react. This distinction, most visible in the large width of the 48 kHz sampling frequency reproduction, is audibly apparent in transient events as a ringing or bell-like sound.

[SmokinJoe]

At the risk of oversimplifying and adding to the flamewar...

As I understand it, virtually all devices use a similar sigma-delta A/D converter type device/algorithm, so a D50, or MR1, or an R09 or a V3 all behave similarly up to a point.  The difference is what happens after that.  On PCM devices, it then takes this 1 bit output and runs it through their own variety of DSP to dither/resample to whatever format you chose in the menu (16/44 or 24/96, etc).  The Korg writes the raw DSD data to disk, and then you do the PCM conversion on your computer.  Whether the Korg software does a better job at home than the Sony/R09/V3 does in hardware/firmware is anyone's guess.

[Nick's Picks]

I never quite figured that out in the A to D chain.
the sigma -delta part is a 1bit stream that somehow makes it to 24 or 16 at whatever sample rates we choose on said hardware.
the whole point of DSD was a new approach to digital that cut out everything after the sigma/delta part and went with that output stream direct into its own formula.
either way,    I like them both.


from Wikki:
A Sigma-Delta ADC (also known as a Delta-Sigma ADC) oversamples the desired signal by a large factor and filters the desired signal band. Generally a smaller number of bits than required are converted using a Flash ADC after the Filter. The resulting signal, along with the error generated by the discrete levels of the Flash, is fed back and subtracted from the input to the filter. This negative feedback has the effect of noise shaping the error due to the Flash so that it does not appear in the desired signal frequencies. A digital filter (decimation filter) follows the ADC which reduces the sampling rate, filters off unwanted noise signal and increases the resolution of the output. (sigma-delta modulation, also called delta-sigma modulation)

The process of creating a DSD signal is conceptually similar to taking a 1-bit delta-sigma analog-to-digital (A/D) converter and removing the decimator, which converts the 1-bit bitstream into multibit PCM. Instead, the 1-bit signal is recorded directly and in theory only requires a lowpass filter to reconstruct the original analog waveform. In reality it is a little more complex, and the analogy is incomplete in that 1-bit sigma-delta converters are these days rather unusual, one reason being that a 1-bit signal cannot be dithered properly: most modern sigma-delta converters are multibit.
Because of the nature of sigma-delta converters, one cannot make a direct comparison between DSD and PCM. An approximation is possible, though, and would place DSD in some aspects comparable to a PCM format that has a bit depth of 20 bits and a sampling frequency of 96 kHz [2]. PCM sampled at 24 bits provides a (theoretical) additional 24 dB of dynamic range.

bla bla bla...., and they wrap it up with
"Sound quality of DSD vs. (24-bit/176.4 kHz) PCM

When comparing a DSD and PCM recording of the same origin, the same number of channels and similar bandwidth/SNR, some still think that there are differences. A study conducted at the Erik-Thienhaus Institute in Detmold, Germany, seems to contradict this, concluding that "hardly any of the subjects could make a reproducible distinction between the two encoding systems. Hence it may be concluded that no significant differences are audible". [4]"

[craig_c]

This is for those who have not yet seen it, here is a link to Korg's mini white-paper on DSD, what they call "Future Proof Recording".
http://www.korg.com/services/products/mr/Future_Proof_Recording_Explained.pdf

From the white paper:
There is no doubt that current 24-bit / 192 kHz audio sounds very good. But there are still areas that can be improved, and approached from a different perspective with other unique benefits.
The Multi-Bit PCM Encoding Process
• A/D & D/A conversion is done in low-bit (e.g.1-bit) at high sample rates, then one in X samples is kept (the decimation process)
• Decimation occurs during recording (i.e. A/D conversion) then interpolation and sigma-delta modulation during playback (D/A conversion)
Surprisingly, most current 24-bit converters actually use 1-bit conversion at the front end already. After capturing a high-speed 1-bit stream the converter uses what is called a Decimation Filter to change the 1-bit data into the desired multi-bit format.

It goes on to explain, in laymans terms, the difference between DSD and PCM.
The upshot is that, like some have presented previously, most PCM recorders DO capture 1-bit at the front end, and immediately process into a programmed PCM format. There is a nice little diagram showing the decimation filter process. Again, this is for us laymen, not you professional engineers who obviously already understand this process and its shortcomings. I'm not advocating DSD, just trying to elucidate for this dialog.

Craig

[headroom]

I asked my personal HIFI GURU about DSD

DSD drawback:
There is not one Programm that can fade in/out in DSD native, no EQ nada. Only PCIe Solutions with Hardware Software and expensive.

1 bit AD Proccesor
There is no such thing possible, it`s a marking lie. One bit or zero bits with a processor gives in the real world eighter 0% Distortion and no signal, or 100 % Distortion and 100% Signal.  1 bit Process is in reality a 4 bit sampling and therafter convertet in PCM or 1 Bit DSD. 1 bit gives maximal 6db  S/N.

In DSD  the Feeback loop of the DSD Signal has X-Grades for AD and DA.
The whole DSD Playback Noiseshaping Process is a constant up and down between Impulse and Noisesuppression  not a help for good sound.

DSD Is a lot more processed and complicated the PCM when you look closer in the real process. DSD is simpler then PCM this is only Marketing blaba.

[headroom]

Audio Engineering Society Convention Paper 5395

Why 1-Bit Sigma-Delta Conversion is Unsuitable for High-Quality Applications
by Stanley P. Lipshitz and Prof. John Vanderkooy Audio Research Group, University of Waterloo

Top Audio Engineers worldwide brings Ultimate Nails for the Coffin DSD/ SACD

http://sjeng.org/ftp/SACD.pdf

Professional Musicians with trained ears have swichted to SONY PCM 50 after the listening expirience with KORG recorders

[Shadow_7]

May of 2001 really?  My device is dated 2006-ish and does 2x's the sampling rate that paper is based on.  Most of thier distortions being above the kHz that's even reproduceable on conventional speakers.  Is it theoretical perfect, I guess not.  Is it better (at 1GB per 12+ minutes, versus 1GB per 1+ hours for CDs), it is IMO.

I suppose that if you're feeding a DSD recorder with the line output of a CD player, no it's not.  Since the source material is effectively the same as a CD recorder.  And a lot of microphone preamps limit their Hz ranges to not make use of the frequencies that could be sampled at DSD bitrates.  But I like my DSD it's clearly better sounding IMO.

I suppose that if the paper was less biased, I might have read it more thoroughly.  But PCM is infinitely perfectible and the opposite argument for DSD.  Clearly biased, i.e. someone with an agenda.  DSD is just another file format IMO.  Maybe there's a lot of junk in that bitrate, maybe there's not.  Once the editing software catches up I guess we'll actually know and not deal in theories.  And audiogate has made some pretty good strides recently.  (if only they had a linux version)

[H₂O]

I also don't want to jump into this "bon fire" of swinging fists over different ways to skin a cat, but personally I ran my Korg MR-1 against my Sound Devices 702 and prefered the sound of the MR-1 - ended up selling my 702 -> I have external Pre's that I prefer much over the internal 7XX pres so the ALL-IN-ONE factor was not a deal with me.  Compared - 24/96 on the 702 and 1 bit converted to 24/96 on the MR-1.
 
FWIW - 7XX's use a more advanced (i.e. a multibit Delta/Sigma) A/D versus standard 1-bit Delta/Sigma A/D such as the Burr Brown models.
 
 

[craig_c]

I'm not sure why this has turned into a debate over whether DSD is a worth a s**t, but clearly at the high end of recording a lot of engineers and studios are using it.
Reference:
Tasacam DV-RA1000HD
EMM Labs
Sonic Studio
Phillips
Sony
Meitner
Sonoma


Craig

[headroom]

For shure DSD Recordings are good but not High End even when the big shots are involdeved
The blind test result speakes for themseself DSD Recordings versus CD most people prefer the CD, someting is wrong. People cannot hear it, because of low Audio chain resolution and/or it is DSD system inherent bot oossible to hear it.

Audio Asylum Thread Printer Hi-Rez Highway SACD just a few comments

A common conception about SACD is that a 1 bit system needs to sample at a rate sixteen times faster than a 16 bit system. However, the SACD 1 bit system is not a weighted system. For example, common rs-232 transmission is via 1 bit, but they are arranged in groups of 8 that make up a byte. Each bit is weighted as a power of 2.
The SACD bit can be thought of as an input to a counter. If the bit is 1 you count up. If the bit is 0, you count down. Therefore, it would take 256 1's to count up to 256, not 8 1's. To count to a 16 bit number takes 65535 1's.
So, in short, the SACD sample rate must be 65536 times the 16 bit sample rate. The equivalent of 44.1 KHZ at 16 bits is 2.8 Giga Hertz at 1 bit.
Or to look at the inverse, at SACD's 2.8 Mega Hertz rate you can describe a 20 KHz signal with 2,800/20 or 140 levels. This is less than the range of 8 bits.
This is not to say that the 1 bit approach doesn't have better low level resolution.

The s/n ratio from scad players above 12kHz is generally lower than that of a 16bit cd player due to the rise in noise floor from 8000Hz and higher.
The signal change rate coming from the DA converion has a finite resolution so sometimes the resulting amplitude is a bit to high and after another 5 or 7 bits it's a bit to low and so on and so on. 
This is the way how the original signal wave is closely reconstructed. 
The little over and undershooting happening during the wave reconstruction is the cause for the 'dsd' noise.
This is all a very cunning balancing act to keep the dsd noise just below the human hearing threshold.

[it-goes-to-eleven]


The SACD bit can be thought of as an input to a counter. If the bit is 1 you count up. If the bit is 0, you count down. Therefore, it would take 256 1's to count up to 256, not 8 1's. To count to a 16 bit number takes 65535 1's.
So, in short, the SACD sample rate must be 65536 times the 16 bit sample rate.

Interesting.  I wondered about that, and was thinking about how it compared to pcm encoding.

I think there is a lot of hype around dsd (and, in the past, marketing dollars).  But the lack of playback options, and the tight control of the format, keep it from getting fairly evaluated.  Especially with unprocessed live material.

[ghellquist]

And again headroom throws a lot of theory. I cannot emphasize enough that a little knowledge is very dangerous. And in these cases each and every theory has to be subject to the real world tests. There is also a problem that headroom draws conclusions that not really can be drawn from the theory I will not go into a pissing contest here, but I suggest people to read his posts with a good sprinkling of salt.

One user, h20, has reported liking the MR1 in tests against a 7xx. This is a valid test based on personal experience. I have done a similar test with my 722 against a MR1000 running both internal and external preamps - I continue to use the 722 based on personal taste and convenience of everything in one package.

// gunnar

[headroom]

Yes yes all theory and snafu

http://www.stereophile.com/thefifthelement/1104fifth/index2.html

PCM vs DSA aka SACD
Again, any processing of the 1-bit stream on the user's end assumes that it will be available at the output of the SACD player. For copyright reasons, this is something that might never be made available.

What is clear is that the Sony/Philips Super Audio CD system does not perform to the same level as the best LPCM converters now available. The best ADCs' delta-sigma modulators, for example, produce four times the data per second as DSD, which means a lower noise floor and lower levels of distortion than can be produced with DSD (assuming the same loop-filter order)

[headroom]

Brainfood for non biased people

http://sound.westhost.com/cd-sacd-dvda.htm

[it-goes-to-eleven]

Yes yes all theory and snafu

http://www.stereophile.com/thefifthelement/1104fifth/index2.html

But what's your point there?  That article just documents that a particular SACD release is apparently the 16 bit pcm audio run through a DSD converter..  And while completely bogus, that says more about the company than about DSD.  Though it does appear to be a case of trying to cash in on dsd hype.

And if the original master is PCM, I don't see the point of converting it to dsd.

[Shadow_7]

Numbers games.  I used to be good at those.

16 bit * 48000 Hz * 60 seconds * 60 minutes == 2,764,800,000 bits per hour
/8 (bits to bytes) == 345,600,000 bytes per hour ~= 350MB per hour
*2 (mono to stereo) ~= 700MB per hour (sounds about right)

Assuming Hz as a per second multiple.  Not all that up on the techno babble.

((24 bit * 96000 Hz * 60 seconds * 60 minutes) / * 2 ~= 2GB per hour

-----

Going backwards based on resulting files.

12 minutes 41 seconds == 1,074,266,204 bytes (DSD @ 5.6MHz)
according to audiogate and windows explorer respectively.
DSD is supposed to have more dynamic range than 24 bit PCM.  So lets call it 32 bit (or not).

12*60 + 41 == 761 seconds

1,074,266,204 bytes / 761 seconds == 1,411,650 (and some change) bytes per second.

1,411,650 * 8 == 11,293,200 bits per second

11,293,200 / 2 == 5,646,600 for mono (5.6MHz imagine that)

5,646,600 / 32 bit == 176,456 (and some change).  Assuming 32 bit, still > 96kHz by the looks of it.

5,646,600 / 24 bit == 235,275 (even).  ~= 2.4 times the rate of 96kHz.  still greater than 192kHz.  But there are PCM devices that do 384kHz now.  And I'm not entirely sure of the bits spec for DSD when converted.  24 bit has greater dynamic range than 16 bit.  DSD according to the Korg manual has greater dynamic range than 24 bit, so anybodies guess.

Stiill 200kHz is better than 96kHz?  At least in terms of bits.  Having a mic and source that uses that bandwidth?  Does a PA system really use it?  Versus something more acoustic like a stringed instrument or brass section?  Does it matter if your mic or preamp doesn't send bits > 48kHz down the line?  Can we expect a new line of HD microphones in our future?  0Hz to 192kHz versus 20Hz - 20kHz?  Who knows a lot of it IS just theory.  In terms of ADC / DAC, my Korg is the best I've got right now.  It's converters are tops IMO.  Even if it's not dealing in DSD.

[craig_c]

With my Korg MR-1 I can record in DSD and using AudioGate convert to 192/24 for editing in DAW. Even with all the math, rounding errors, SigmaDelta, Foucault transcription errors and quantum interference from n-dimension strings, (pardon my sarcasm, or not), isn't it pretty damn likely that I will have a more usable, higher quality audio file, than if I had recorded with the Sony PCM D-50 at its max of 92/24, and used that as my "RAW" file to edit in the DAW?

Craig

[it-goes-to-eleven]

isn't it pretty damn likely that I will have a more usable, higher quality audio file, than if I had recorded with the Sony PCM D-50 at its max of 92/24, and used that as my "RAW" file to edit in the DAW?

I think that would be an overly optimistic assumption, with many variables.  It also depends on how you define "higher quality".  Is that what sounds most pleasing to the ears?

If you asked me whether the 24/192 on the 7xx is better than 24/96 on the 7xx, I wouldn't necessarily assume so.  I'd say you need to test it.  But at least you'd be comparing the same line in, with the same impedance, noise, etc.

How are you getting the output from your contact mics into the recorder?  Have you done any testing with your mr1?

[craig_c]

"Higher Quality"... on the one hand I think it is faithfulness to the sound with minimal noise, i.e. "fidelity". But perhaps it IS more at "pleasing to the ears".

For my sampling project "higher quality" means it hasn't been stepped on a lot, so that when I downsample the file I don't get a lot of whooshing, phase shifting, wow-ing.

I think the problem I have is one of taking an analogy too literally. I'm a photographer, a Photoshop specialist for 18 years. I work with RAW files all the time. In my work "higher quality" means "accurate color", "wide gamut color space", "high resolution" (without chroma fringing), "in focus", "proper exposure". The better my RAW file is to start with, the less degrading all of the post production will be to the image. If I have to take a very dark image and process it to pull out detail in the shadows, I may end up with very noticeable banding - sudden jumps in tone value. 

SO, now I'm starting to play with audio. I have been assuming if I start with an audio file that has a much higher frequency and bit-depth than Redbook, when I start to radically resample that file in my DAW that artifacts from that pitch shift will be minimized. What I'm starting to see is that the photo analogy that Korg uses in its "Future Proof Recording", i.e. suggesting that the DSD file is like a RAW photo file, may not be accurate in the way I was thinking about it. This is a kind of category error.

There is a way that the analogy works, but perhaps not to the extent I was thinking of it. -- If DSD moves noise into the 20-24KHz range, and I'm taking a PCM version of that file and shifting the frequency down a lot, then I'll probably be moving noise right down into the range of human hearing.  -- Must test more.

As to the piezo pickups, I'm running them straight into the mic input. They have a surprisingly strong signal. The difficulty I'm running into is finding an adhesive material that makes a very strong contact with a surface, but then will release without destroying the pickup in the process.

Craig

[headroom]

I asked a few friends about Internet Treads and more....
In the last five years the quality posted in Treads was sinking rapidly, but the amount of words went up. The amount time to follow is eating more and more time and the content is tinner and tinner. Sometimes just stubbern poeple wich did not read or understand the matter takes a lot of space. But with quick replys within minutes - not even think for themself, what was the inherent information. Repeated mantras of belive systems collides.  Some of them want shorcuts instead of going deeper. And o lot of people are in silence still reading all, can hopefully profit  from following it. I think the US Buddy princip (helping from the cracks) in Forums is very good and seldom in Europe. Hope you can take this not to personal.
I act sometimes just as signpost ,sometimes with own arguments/expiriences, infos from musicians wich make recordings for themself. Not looking for a longest penis contest. Hope can help a bit. We can all learn from each other, expanding the understanding and make better recordings.

[headroom]

And here are some briiliant audio masters in deep discussion again...

Post Nr 21

http://www.gearslutz.com/board/remote-possibilities-acoustic-music-location-recording/89367-new-korg-portable-dsd-recorders.html

[headroom]

Removing aliasing effects in the 20khz-24.1khz band takes care of digital "glare." How would this be accomplished? Better brick-wall filters? Aren't those aggressive digital filters part of the problem?

The answer is in this pdf

[Todd R]

isn't it pretty damn likely that I will have a more usable, higher quality audio file, than if I had recorded with the Sony PCM D-50 at its max of 92/24, and used that as my "RAW" file to edit in the DAW?

Craig

If you are a photographer then maybe this analogy will help:

I've got an original Canon digital rebel, with a 6.7MP resolution, among other things.  Say you've got a brand new Canon 50d with 15MP resolution, and a whole lot other improvements over my digital rebel.

I'm shooting with the most expensive Canon L glass that cost thousands.  You're using a 20 year old cheapo Tamron lens you got from Ebay that probably has fungus growing in it.  Isn't it obvious that your photographs will be more high quality and true to life than mine since your camera's sensor is so much better?

Well, perhaps, but I wouldn't even say yes, let alone say it is pretty damn likely.  I have no idea how the analog front end inside the MR1 compares to the analog front end of the Sony D50, but it is a huge leap to say that all of that doesn't matter and the only thing that matters is the DSD conversion vs the PCM conversion.

[headroom]

isn't it pretty damn likely that I will have a more usable, higher quality audio file, than if I had recorded with the Sony PCM D-50 at its max of 92/24, and used that as my "RAW" file to edit in the DAW?

Craig

If you are a photographer then maybe this analogy will help:

I've got an original Canon digital rebel, with a 6.7MP resolution, among other things.  Say you've got a brand new Canon 50d with 15MP resolution, and a whole lot other improvements over my digital rebel.

I'm shooting with the most expensive Canon L glass that cost thousands.  You're using a 20 year old cheapo Tamron lens you got from Ebay that probably has fungus growing in it.  Isn't it obvious that your photographs will be more high quality and true to life than mine since your camera's sensor is so much better?

Well, perhaps, but I wouldn't even say yes, let alone say it is pretty damn likely.  I have no idea how the analog front end inside the MR1 compares to the analog front end of the Sony D50, but it is a huge leap to say that all of that doesn't matter and the only thing that matters is the DSD conversion vs the PCM conversion.

 I see it like this as analogy DSD has a coating without UV Filter and ghosting, less contrastpicture then a PCM lens, Because DSD have lesser resolution in the higher frequencys (Blue UV)

There is also another essential difference; the increase in resolution you achieve from raising the sampling frequency will be frequency dependant. A one-bit system will therefore have high resolution at low frequencies (where the information theoretically is low) and have low resolution at high frequencies (where the information theoretically is high).

By the use of noise shaping of high order, it is possible to increase the resolution at "quite high frequencies" at the expense of resolution at very high frequencies, but only for static, non transient signals. Transient signals will have poor resolution in a one-bit system. If the signal does not endure for a long enough time, the error will not be minimised by the noise shaper of the one bit system.

That's why you can read in documents from Burr Brown (who manufactures both one-bit and multi-bit converters) that you should use multi-bit converters for "waveform synthesis applications requiring very low distortion and noise". They have not written this for nothing.

A one-bit converter (i.e. the DSD system) cannot regenerate a short pulse with stringent form. It will change form from moment to moment. Every identical recorded pulse will show up with a new form.
Just to avoid misunderstandings I want to make a reminder that SACD has higher resolution below 5-10kHz than the CD-system. Exactly where the limit is, where each system (CD or SACD) is better, depends on if you are looking at a static or dynamic signal.

At frequencies below 100-600Hz the SACD-system could theoretically be even better than DVD-Audio, but in reality this is not important. We are talking about so small flaws, far below the hearing threshold, so they can be disregarded. Any specific player however, can be very bad at low frequencies, but not due to the system if SACD or DVD-A is used.
The noise from SACD just above 100kHz is higher in level than most of the treble in the audible range, at least when listening to the majority of acoustical music. It can also be discussed if DSD uses a "high" sampling rate. But apart from that: Yes, without noise shaping it will not work at all and that would be a lot worse. Now it is only a little bit worse than CD in the highest treble.

But why introduce a new super high-resolution system, that is "a little bit worse" than CD? Of course there are advantages when compared to CD also, but the drawbacks of SACD/DSD are completely unnecessary. Shouldn't a new system be better than CD in all aspects?

The noise level in the range above 100kHz is –40dB under maximum signal level (and is thus even visible on an oscilloscope!). The noise is in fact much higher than any possible music signal in the same frequency range. This can be compared to DVD-A where the noise level is –144dB in the whole audible range and also in the ultrasound range.

[Shadow_7]

The noise level in the range above 100kHz is –40dB under maximum signal level (and is thus even visible on an oscilloscope!). The noise is in fact much higher than any possible music signal in the same frequency range. This can be compared to DVD-A where the noise level is –144dB in the whole audible range and also in the ultrasound range.

Isn't DVD-A just 24/96?  aka ~2GB per hour, where DSD at 5.6MHz is ~5GB an hour of data.  Not to say that the lenses are equal.  But what < $1K speaker reproduces 100kHz?  Does the DAC on an iPod even try to produce that?  My best headphones cut out at 35kHz.  And most humans hearing cuts out at < 25kHz.  Who cares what's 100kHz and up?  Bat scientists?  Okay so DSD isn't best suited for bat science / research.  It still makes a pretty good CD and has a decent amount of dynamic range for those who like recording at > 16 bit.

[headroom]

The noise level in the range above 100kHz is –40dB under maximum signal level (and is thus even visible on an oscilloscope!). The noise is in fact much higher than any possible music signal in the same frequency range. This can be compared to DVD-A where the noise level is –144dB in the whole audible range and also in the ultrasound range.

Isn't DVD-A just 24/96?  aka ~2GB per hour, where DSD at 5.6MHz is ~5GB an hour of data.  Not to say that the lenses are equal.  But what < $1K speaker reproduces 100kHz?  Does the DAC on an iPod even try to produce that?  My best headphones cut out at 35kHz.  And most humans hearing cuts out at < 25kHz.  Who cares what's 100kHz and up?  Bat scientists?  Okay so DSD isn't best suited for bat science / research.  It still makes a pretty good CD and has a decent amount of dynamic range for those who like recording at > 16 bit.

You miss the point, it is about the noise in that region not the music signal. This high frequency noise  intermodulate the audible signal and this is one problem. The other is, that this noise is not static, the strong feedback loop DA / DSD is not the cure.

[Shadow_7]

The noise level in the range above 100kHz is –40dB under maximum signal level (and is thus even visible on an oscilloscope!). The noise is in fact much higher than any possible music signal in the same frequency range. This can be compared to DVD-A where the noise level is –144dB in the whole audible range and also in the ultrasound range.

Isn't DVD-A just 24/96?  aka ~2GB per hour, where DSD at 5.6MHz is ~5GB an hour of data.  Not to say that the lenses are equal.  But what < $1K speaker reproduces 100kHz?  Does the DAC on an iPod even try to produce that?  My best headphones cut out at 35kHz.  And most humans hearing cuts out at < 25kHz.  Who cares what's 100kHz and up?  Bat scientists?  Okay so DSD isn't best suited for bat science / research.  It still makes a pretty good CD and has a decent amount of dynamic range for those who like recording at > 16 bit.

You miss the point, it is about the noise in that region not the music signal. This high frequency noise  intermodulate the audible signal and this is one problem. The other is, that this noise is not static, the strong feedback loop DA / DSD is not the cure.

What exactly is to be cured?  Data that's going to be effectively thrown out for all practical purposes?  A little like washing your trash before taking it to the curb, don't you think?  It's not like any of the other edits we do in post is any less destructive.  MP3 anyone?