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Audio @ 1 bit 5.6 MHz on Blue Ray could save the day... Recorders, Players & Tape Machines
Old 26th June 2009
  #91
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I need to note that remarkably DSD on Blue Ray disc has yet to save anyones day.

Again -
Is DSD a nice option for a different vibe for high resolution small track count recording that can appeal to a small niche of dedicated recordists? Absolutely.

Is there any way that the recording industry can regain the level of profitability that it had a few years ago through delivery of DSD recordings to the end listener? No way in heck.

I continue to strongly feel the best hope for the continuation of fidelity into the future is through pushing download distribution of high res PCM in a lossless compression codec - with the best case scenario of the format that gains most use being open source and license free such as FLAC. DSD is simply too large in its file sizes, too patent encumbered, and too dependent on specialized DAC's, for it to make sense as a consumer format except as a small niche product. As a small niche product I think it has its place - it's just absurd to think that it will ever get beyond this.

Best regards,
Steve Berson
Old 26th June 2009
  #92
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Quote:
Originally Posted by Cellotron View Post
I need to note that remarkably DSD on Blue Ray disc has yet to save anyones day.

Again -
Is DSD a nice option for a different vibe for high resolution small track count recording that can appeal to a small niche of dedicated recordists? Absolutely.

Is there any way that the recording industry can regain the level of profitability that it had a few years ago through delivery of DSD recordings to the end listener? No way in heck.

I continue to strongly feel the best hope for the continuation of fidelity into the future is through pushing download distribution of high res PCM in a lossless compression codec - with the best case scenario of the format that gains most use being open source and license free such as FLAC. DSD is simply too large in its file sizes, too patent encumbered, and too dependent on specialized DAC's, for it to make sense as a consumer format except as a small niche product. As a small niche product I think it has its place - it's just absurd to think that it will ever get beyond this.

Best regards,
Steve Berson
Points well taken Steve. I do feel that there has been a missed opportunity with this format. I do however feel that physical product is very important. For me I really like having my music collection backed up with my cd's. I have had hard drives die on me in the past and that would be very bad if someone had all their music on their hard drive with no back up. Plus, I think that the industry should implement something similar to what UAD has done with plug ins. In other words, hypothetically you should only be able to play the 1 bit 5.6 MHz discs if you have an approved player and the disc is physically in the player. No more of this file sharing crap, the pirating has got to end!

DSD sounds quite a bit better to me at 1 bit 5.6 MHz. So if it was implemented correctly and skilfully introduced on Blue Ray I think it could have been a great thing. This opportunity has seemed to have been missed by Sony for some reason though. Either way, personally I love recording on it and think that 1 bit 5.6 MHz DSD has no current parallel in the digital world to my ears. It's the closest to high end analog recording that I've come across.
Old 26th June 2009
  #93
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Quote:
Originally Posted by oky**** View Post
hi,

same deal. signal adds linearly, quantization noise adds with a square root.

faster, pussycat.

right.
Not in my PCM
Old 26th June 2009
  #94
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Quote:
DSD sounds quite a bit better to me at 1 bit 5.6 MHz.
I'm a bit surprised to hear this. Are they using fixed crystal clocks? Most cheap recording devices like that use analogue voltage controlled oscillators for clocks. That means, the higher you clock them, the higher the jitter. I have a couple of converters here that sound their best at 48KHz! But then, I have another set that sounds best at 96KHz because it has a fixed clock and simply down-samples for lower rates.



Quote:
So if it was implemented correctly and skilfully introduced on Blue Ray I think it could have been a great thing.
The Blu-Ray standards were set in 2002. THEY WILL NOT CHANGE THEM! Not for any reason, it costs too much.

I agree that a psysical product is important. I'll be the first to admit that I make illegal downloads all the time. I do this to sample the product before comitting. Just last week, I downloaded an album from the early 90s. I liked it so I went to a store and bought it. I also downloaded U2's latest record and it didn't impress me so I deleted it. Almost all of the albums I've bought over the last 5 years have been because I could find them and hear them on the internet. I've gotten stuff I'd never normally be interested in getting because it was so easy to find them and hear them.
Old 26th June 2009
  #95
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Quote:
Originally Posted by wado1942 View Post

The DBX-700 (which used adaptive 1-bit 700KHz) converter got about 19-bit performance in the mid-80s taking up about the same space as 16-bit 44.1K PCM.

A bit less (no pun intended) according to the specs given here: dbx Model 700 Digital Audio Processor - Wikipedia, the free encyclopedia

Quote:
Technical specifications

  • Dynamic Range: 110dB typical with A-weighted noise 20Hz-20kHz; >105dB unweighted
  • Frequency Response: 20Hz-20kHz, sine or pink noise, 100mV, reference record position
  • THD: less than 0.05%, 1V input, 1 kHz
  • Wow and flutter: less than 0.01% unweighted; 0.006% wrms
  • Anti-aliasing filters: -3dB at 37 kHz
  • Sampling Rate: 644 kHz
  • Bit Rate: 644 kbit/s
  • Mic Pre: adds less than 1dB noise, 100 to 1k-ohm impedance
Alistair
Old 27th June 2009
  #96
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Quote:
Originally Posted by bcgood View Post
For me DSD/WSD @ 1 bit 5.6 MHz sounds so much better than PCM it's scary.
Then, no offense, your DA converter sucks.

PCM audio's last frontier is simply two things: filtering, and bit-depth. We have the bit-depth covered at 24bit, but 32bit wouldn't hurt for the final standard... until humans evolve much better hearing.

Reconstruction filtering on the other hand can still come a LONG way to mathematically perfectly re-creating the original sound. Many converters out there now "cheat" by upsampling everything to a much higher rate because of the inadequacy of the filter technology we have now, for the most part.

Eventually there will be a time when the filtering approaches perfection, and we can finally realize Nyquist's theorem:
The sampling theorem asserts that the uniformly spaced discrete samples are a complete representation of the signal if this bandwidth is less than half the sampling rate.
in the DAC's output as well.

DSD will never be capable of a complete (mathematically perfect) representation, by design.
Old 27th June 2009
  #97
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Quote:
Originally Posted by Sounds Great View Post
That is all fine, but it doesn't explain why DSD sounds so much better than PCM.
No but placebo does.

Seriously though, your comment is meaningless as it stands. Are you comparing DSD and PCM signals on the same converters? And if so have corners been cut on the PCM implementation? (This could very well be done to promote DSD over PCM. Much worse things have been done by companies).

In other words, without qualifying your comment and demonstrating that this is really true (with proper double blind ABX testing) it is useless.

Alistair
Old 27th June 2009
  #98
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Quote:
the converters in the F1 were more linear.
I had the 501 and 601 for mixdowns, BITD, but have a slightly off topic question concerning the hifi audio tracks of a "hifi" vcr. I remember around the same time as F1, I used to mix down to the hifi tracks of a "hifi" vcr which sounded close to the 501. Were those pcm or just very clean analog? I remember the quality being pretty decent.
Old 27th June 2009
  #99
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Jesse Graffam's Avatar
 

Quote:
Originally Posted by wikipedia
VHS Hi-Fi audio is achieved by using audio frequency modulation (AFM), recording each of the 2 stereo channels (L, R) on a frequency-modulated carrier and embedding the modulated audio signal pair into the video signal. To avoid crosstalk and interference from the primary video carrier, VHS's implementation of AFM relied on a form of magnetic recording called depth multiplexing. The modulated audio carrier pair was placed under the luminance carrier (below 1.6MHz), and recorded first. Subsequently, the video head erases and re-records the video signal over the same tape surface, but video signal's higher center frequency results in a shallower magnetization of the tape, allowing both the video and residual AFM audio signal to coexist on tape. (PAL versions of Beta Hi-Fi use this same technique). During playback, VHS Hi-Fi recovers the depth-recorded AFM signal by subtracting the audio head's signal (which contains the AFM signal contaminated by a weak image of the video signal) from the video head's signal (which contains only the video signal), then demodulates the left and right audio channels from their respective frequency carriers. The end result of the complex process was audio of outstanding fidelity, which was uniformly solid across all tape-speeds (EP or SP.) Since JVC had gone through the complexity of ensuring Hi-Fi's backward compatibility with non-Hi-Fi VCRs, virtually all studio home video releases contained Hi-Fi audiotracks (in addition to the linear audio track).
neat eh?
Old 27th June 2009
  #100
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Quote:
Originally Posted by uncajesse View Post
neat eh?
I remember the quality being pretty amazing and a lot less expensive than open reel. but never new how it was accomplished...
Old 27th June 2009
  #101
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Quote:
I used to mix down to the hifi tracks of a "hifi" vcr which sounded close to the 501. Where those pcm or just very clean analog?
HiFi VHS is FM encoded into the same space on the tape occupied by the video using the spinning drum heads. So frequency content is the only factor in determining the playback quality. As long as the input amplitude is low enough to avoide over modulation, that is. If the input signal is too hot, it'll saturate the modulator and the video/audio signals interfere with each other. I used to do the same as you, even up to 4 or 5 years ago when I did mixes at other studios, before I got the 1/4" deck.



Quote:
DSD will never be capable of a complete (mathematically perfect) representation, by design.
Neither can PCM. Both can get ever closer, but never match the source.



Quote:
Reconstruction filtering on the other hand can still come a LONG way to mathematically perfectly re-creating the original sound.
Yes it can. But as I said, it will never be perfect.



Quote:
Many converters out there now "cheat" by upsampling everything to a much higher rate because of the inadequacy of the filter technology we have now, for the most part.
That's not cheating. THAT'S HOW THE FILTERS WORK! You interpolate new spline-fitted samples between the existing samples, pushing the out the percieved error by 8x usually, then rolling off the extreme high end in the analogue domain to reduce interference. Personally, I'm waiting for the day we can just use 352.8KHz linear and won't need downsampling for storage. Then we won't need digital anti-aliasing filter on ADC or reconstruction filters on DAC. Then we won't have to have plugins that double or quadruple the sample rate internally.
Old 27th June 2009
  #102
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Quote:
Originally Posted by wado1942 View Post
That's not cheating. THAT'S HOW THE FILTERS WORK! You interpolate new spline-fitted samples between the existing samples, pushing the out the percieved error by 8x usually, then rolling off the extreme high end in the analogue domain to reduce interference.
Spline interpolation is rarely, if ever, used in audio. Also not "perceived error" just an intrinsic part of the sampling process.

What actually happens is that zero values are inserted between the original samples until the desired sampling rate is generated, then a digital filter is used set to the new Nyquist frequency.

Quote:
Personally, I'm waiting for the day we can just use 352.8KHz linear and won't need downsampling for storage. Then we won't need digital anti-aliasing filter on ADC or reconstruction filters on DAC. Then we won't have to have plugins that double or quadruple the sample rate internally.
Why? What would be the advantage over the 6 Mhz multi-bit quantizers in use today?


DC
Old 27th June 2009
  #103
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Quote:
Originally Posted by dcollins View Post



Why? What would be the advantage over the 6 Mhz multi-bit quantizers in use today?


DC
hi,

david, i'm no expert, but i think it is reasonable to say that it would be better to not have to apply a bunch of interpolation, decimation, and so forth to get a 24 bit result, no?

all that stuff is prone to error in varying degrees. the more "stuff" you have to do the, the weirder it is.

also, the more stuff, the more latency. that's a big reason to do less stuff, in my opinion. i note that engineers sometimes don't put as much importance on latency as musicians do.


right.
Old 27th June 2009
  #104
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Quote:
Originally Posted by Waltz Mastering View Post
I remember the quality being pretty amazing and a lot less expensive than open reel. but never new how it was accomplished...

hi,

i have one of those decks around somewhere, also. i remember keeping it because the audio was better than a lot of other stuff.

they don't make 'em like they used to?



right.
Old 27th June 2009
  #105
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I have a JVC Super VHS that I sometimes use to track and mix to. It records analog audio at extremely high fidelity. It sounds really BIG and smooth. There where some VHS hi-fi and Super VHS recorders that recorded in PCM but not many. Basically you'd have to check your manual to see if it recorded that way, it was mostly Panasonic that utilized this method towards the end of their production.
Old 28th June 2009
  #106
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Quote:
Originally Posted by wado1942 View Post
Personally, I'm waiting for the day we can just use 352.8KHz linear and won't need downsampling for storage. Then we won't need digital anti-aliasing filter on ADC or reconstruction filters on DAC. Then we won't have to have plugins that double or quadruple the sample rate internally.
I wonder if that chipset I mentioned above is capable of 352.8kHz as well as 384kHz. I'll have to look through the technical specs later on. I really do want to mess around with one, but I'm not sure if they sell samples to a simple hobby buyer such as myself. And... I would have to buy VisualDSP++, unless they are providing a license for that with the chipset.
Old 16th March 2011
  #107
Quote:
Originally Posted by bcgood View Post
As far as editing goes if you used a bunch of the Korg units with a contraption to press the play and record buttons simultaneously then you could use the Korg unit's like a multi-track tape deck.
...Rachmaninoff has big hands!

(I love these Korg boxes, too.)
Old 16th March 2011
  #108
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Zombie thread, eh? The big problem every time these threads pop up is the following (from the O.P.):
Quote:
Originally Posted by bcgood View Post
When people hear the sound of 1 bit 5.6 MHz audio they will be blown away, I know I was. If this was deployed and marketed properly and new recordings where recorded for best fidelity and not loudness I think this could actually save the music industry.
"Not loudness". That's the thing. That's what renders this whole wish irrelevant.

Without any upgrade in format or technology, with just the ancient 44.1 16 bit we have right now, we could make a phenomenal improvement in fidelity simply by knocking off the damn loudness. The "not loud" leap in fidelity would absolutely dwarf the "new format" bump in fidelity.

So if it costs nothing at all to blow the fidelity doors off of the world right now and we're still not doing it (because we suck I suppose), why even bother discussing an expensive upgrade that would provide a relatively small improvement?
Old 17th March 2011
  #109
Quote:
Originally Posted by Cheebs Goat View Post
So if it costs nothing at all to blow the fidelity doors off of the world right now and we're still not doing it (because we suck I suppose), why even bother discussing an expensive upgrade that would provide a relatively small improvement?
Uhhhhh...

...

...

...



'cause we're 'slutz?

;~)
Old 17th March 2011
  #110
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Truth is stranger than fiction.

---------------------------------

KORG CLARITY

A CPU-based DAW software is in development (tentatively called Clarity). It runs on a Windows PC and supports ASIO 2.1 DSD mode. When combined with Korg's USB audio interface, it delivers multi-track DSD recording, playback, editing and mixing at both 2.8 MHz and 5.6 MHz sample rates.

Old 17th March 2011
  #111
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zenmastering's Avatar
I heard this system at the last AES in San Fran...I thought it was excellent in both sound and operational stability. Heres' to hoping that Korg actually release this thing...

Graemme


Quote:
Originally Posted by Sounds Great View Post
thumbsup


I could see giving up my dream about having consoles and tape machines again for such a contraption.

--

Still I think the real test will be the results of in the box summing in the dsd domain.
Old 17th March 2011
  #112
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Quote:
Originally Posted by Sounds Great View Post
thumbsup


I could see giving up my dream about having consoles and tape machines again for such a contraption.

--

Still I think the real test will be the results of in the box summing in the dsd domain.
Despite what a lot of people think, tape machines and consoles especially are common place in a studio environment. Also, summing in the DSD domain is an intrinsic impossibility. The advantage of DSD is that it converts back & forth between analogue & pulse density very easily.
Old 17th March 2011
  #113
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That's correct. In fact, there's several DAWs that can handle DSD, but they convert to PCM for the actual processing. I think Korg keeps their actual specs under wraps but another (forget the name) specifies that they convert to 32-bit 384KHz to handle the work. There's some stand-alone DSD consoles that were at least in the works that worked similarly. That's the thing though, just to mix two tracks together, you'd need at least two bits or else there's no headroom to sum.
Old 17th March 2011
  #114
Quote:
Originally Posted by bcgood View Post
LOL I figured I'd get some hate from the establishment. For me DSD/WSD @ 1 bit 5.6 MHz sounds so much better than PCM it's scary. That's really the bottom line. If you're a mastering engineer then hopefully you can hear the difference between the two. Don't underestimate what putting sound quality first could do for the industry...

Cheers
What PCM?
A well mastered CD can still sound better than SACD, as John Watkinson has proved. There is nothing above 23kHz on an SACD except noise, and lots of it.

[IMG=http://img24.imageshack.us/img24/49/sacddtscomparisons.jpg][/IMG]

This comparison is admittedly flawed, but still very revealing.
I took the SACD and the DTS-DVD version of the same song (It cannot possibly be the same master, but that's a whole different kettle of fish) and recorded the decoded output into my DAW.
Both discs used the same chain:
Denon 3910 Universal player decoding internally.
Output direct into RME ADI-8 DS Converters at 24/96 resolution.
Converters into Nuendo via RME HDSP9652 I/O card with all clocking via Wordclock off the ADI-8 DS.

As you can see on the image, the DSD has a greatly reduced dynamic range with the decoded DTS 9624 streams having typically 30-35dB more range.
Additionally, the spectrum plot tells another story, as you can clearly see the ultrasonic noise taking off from around 22kHz. By comparison, the DTS goes flatline, revealing the sampling rate the original mix was done at as probably 44.1kHz.

Other reasons the DSD will definitely sound different - although whether or not it is better in my opinion is certainly open to debate - is because of the process it uses - DSD was developed originally as an archival storage media, not for consumer release, and as a result of it getting turned into such requires a heroic amount of noise shaping.
Encoding a PCM mix to DSDIFF for SA-CD also requires great care to be taken in the mastering, as the most common cause of encoding failure is too much energy in the top end - all too common these days with the loudness wars showing no real signs of an early cease-fire, and this can possibly result in a different master being used for the SA-CD over the Red Book version.
This idea is also hinted at in the spectral plots of the attached image, and again could on it's own counter the argument the DSD sounded "better" - I do not know as I was not there at the mastering session

In closing, I am certainly not going to try & tell you that what you heard was not, in fact, what you heard and that the CD was better. This would be nonsense, as I was not there so I have no idea. All I am trying to say is that DSD is not "the answer" as it were, and indeed has problems of it's own (some would say lots of them).
Yes, I also have an axe to grind as well, as I still author for DVD-Audio so anything I say is likely to be biased - but I did try the SA-CD authoring tools again recently thanks to the good chaps at Sonic, and my position has not changed. If I honestly thought it would be of a benefit to our clients, I would simply buy the kit (it is not expensive these days, considering) and get on with it.

BluRay as well has issues, not the least of which is the cost. Again, this is an area we are getting into though, as I think it will be of benefit for films & concert releases as it offers lossless surround instead of Dolby Digital. As a music form, I am not too convinced as yet because the player penetration just isn't even close to that of DVD so it's almost mandatory to do an "SD" version as well - and this is very clever as it implies the DVD is somehow a lesser version. In DVD-Video terms, this might be true if we are talking about a film release or a concert title but if we are talking pure music then there is no need at all for the DVD to be in any way a lesser product.
In the Video_TS you can have stereo PCM at up to 24/96. Okay, some players will either truncate this to 16-bits or sample rate convert on the fly to 48kHz or even both - but the BluRay picture isn't much clearer as it too was developed by committee.
You can have the 5.1 mixes in DTS 9624, which is very good indeed and whilst it is definitely not lossless it is by no means artifact ridden and a lot of people cannot tell the difference.
We can also put bonus video into the equation too - something a "Pure Audio" BluRay disc cannot do unless you add a full BDMV titleset to it, and then up go the costs across the board. Still score to BluRay here though, as you get the HD footage with lossless surround, so that is a big +1.
The Pure Audio option is also good, but you will need to speak pretty fluent Java to make it happen (or go to Stefan at MSM, of course).

Moving to the Audio_TS, this will give us everything you get in Pure Audio BluRay except 7.1 and 5.1 at 24/192 resolution (your upper limit is 5.1 at 24/96) - without all the hassles of mandatory copy protection systems and if you add a Video_TS to it you get something that will play on every DVD player ever made in one form or another. Sure, it can easily be argued that you need a special player to access the High Resolution section - but you need one of these for BluRay and SACD too.

Turns out it is far cheaper to make a CD/DVD-A/V digipack than a BD/DVD/CD set, and unless you specifically need HD video oe 24/192 5.1/7.1, it adds nothing extra you can't already do on a DVD.
Old 17th March 2011
  #115
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Quote:
Originally Posted by Sounds Great View Post
If the system is to be operating in a PC computer, would the mixing and editing be working in PCM?
Yea, I didn't like that it was on a PC either. I wanted to see a standalone unit like the, MR 2000 with multiple ins and outs that you could then mix on a console such as API etc.
Old 17th March 2011
  #116
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You noted that you understand your comparison is flawed. I'll take it a step further and state that while interesting, it's completely pointless. I believe standard DTS is six channels of 48KHz audio at 1.536mbps (though, often much lower). You can hear all the mush in the surround channels because most of the bandwidth goes to the center channel. It's an obviously inferior format. That said, your experiment just shows that the DACs in your player don't try very hard at rolling off dither noise. For a decade, most PCM recordings were made FROM 1-bit Delta-Sigma streams. It's just that they were encoded to PCM before storage. The DTS source had all that dither noise in it just like the DSD recording you sample. In decimation, that noise was filtered out of the signal. I admit DSD has problems, but it is by nature superior to CD-DA and DTS. While modern converters have up to six bits now (as far as I've seen), they still employ time smearing filters that still don't do nearly enough to avoid aliasing at 48KHz, forget 44.1KHz. DSD avoids those issues by simply not decimating the signal. At 96KHz, it's a much more difficult argument, especially with the advances in ADCs over the last few years. Though, even 96K multi-bit converters still use HEAVY dither & noise shaping. I still think avoiding the need for digital LPFs and heavy-handed reconstruction filters is a good one.



Quote:
Originally Posted by Sounds Great View Post
PCM summing in the box is where most of the sound is lost.
Actually, summing is the one area where PCM processing is completely lossless. You simply add numbers together. No distortion of any kind.
Old 17th March 2011
  #117
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narcoman's Avatar
 

Quote:
Originally Posted by Sounds Great View Post
Hogwash.
it'd a brave, and likely unwise, man to argue with John Watkinson.
Old 19th March 2011
  #118
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It's a common misconception that summing, mixing and other processing are all impossible in 1-bit audio, natively. In fact they are all perfectly possible, and the Sony Oxford console (demonstrated several years ago) does them all.

But 1-bit recording is not an efficient way to use bits. DSD-wide (4 bits) would make more sense but I don't see much sign of it being taken up. And personally I'm happy enough with 44/16 audio.
Old 20th March 2011
  #119
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Laarsø's Avatar
 

Quote:
Originally Posted by bcgood View Post
...run one left and right to the korg recording at 1 bit 5.6 MHz and the other left and right go to a lavry gold or whatever at 24/192 PCM. Now do a double blind ABX test and see which one comes out on top. I know that I could hear the difference, can you, who knows?

As far as syncing the Korg unit's, it is possible. I could post schematics that I've designed to do just that.
OK. But Lavry refuse to implement 4x Fs PCM conversion. Is less accurate with settling time necessity factored in, even though anti-alias filter is further away. Smoother, but less accurate, is unacceptable in (their) digital doctrine.

I did like the sound of DSD surround on PMC's (ironic?) in Vlado's room at Sony. But the sound of analog tape is for me where it's at. Real drum Dub at 30 ips AES, on 1/2" 2-track with no NR and no penthouse using 456 or ATR, depending.




Cheersø,
Laarsø
Old 20th March 2011
  #120
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even better - GP9.....
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