I heard through the grapevine that the transfers of multis that created "love" were done at Metropolis at 96/24, but that's not first-hand.
The answer about stock machines capturing bias:
Ours is purpose built, extended range to 300kHz with minimal rolloff and low noise. It includes a multiplexer that shifts captured bias to 24kHz ( or 48) on the fly. And the audio kills, and it by far yields the most reliable results. Only been stumped by two tapes in 12 years, and either we goofed or the bias was erased by poor demagged heads during its life. Setting that aside.
1.Nyquist limits 192/24 capture of raw bias on any machine to 96kHz.
Some older machines like c37/j36 ca. 80kHz bias but vast majority = or >100kHz
Workaround: 1/2 speed playback
2. Head gaps and electronics limit output on ATR to about 115kHz see above. But it can often yield enough to do a rough job and get results.
3. Studers roll off lower and have an aggressive notch filter at 80kHz to keep erase noise out of the playback. Sometimes that erase freq bleeds into the record head and therefore can be captured on an ATR see above.
4. ATR LCD display bleeds 28.8kHz onto all recordings it makes (and playbacks too).
Playing an ATR recording on a Studer or MCI reveals this signal and it's useful if the recording isn't super hot. Hi-hats and snare transients interrupt it if the recording is too hot.
So on occasion we've been able to rescue or demo with a tape we can't access based on some luck. There was a 15734 sync buzz from a nearby tv monitor that worked well.
The Beatles Sullivan DVD was the same. That's on our Facebook page.
And as noted, all the Beatle's earlier than the 3M (which we could handle) would be likely to be successful with stock ATR 102 192/24 transfers.
But yeah, the only truly reliable method is to lease/buy the system or send the tape to John Chester or Airshow.
Please pardon the misspellings and occassional insane word substitution I'm on an iPhone
> On Aug 29, 2014, at 1:27 PM, Tom Fine <[log in to unmask]> wrote:
> Hi Jamie:
> As I understand it, the last time the Beatles stereo master tapes were transferred to digital, it was at 44.1/24. Those files were FLAC'd and sold on a USB drive. For the stereo CDs, there was some dynamics compression done (Abbey Road engineers claim "only a couple of dB here and there, to give the discs a modern sound level"). The stereo LPs in the recent box set were cut from the 44.1/24 un-compressed files.
> I have read contradictory things about the mono CD box set, but the direct quotes from Abbey Road engineers who actually worked on it, as printed in reputable publications, indicate the same process was used -- 44.1/24 transfers. The mono CDs were claimed to have no dynamics compression. The new mono LP box set coming next month was reportedly made all-analog, no digital files used. Since they re-played the mono master tapes, one hopes they made a high-resolution digital capture at the same time. They seem standardized on 96/24, so that may be as high resolution as they go. Again based on press reports I've read, the mono master tapes are in good playable condition but there are issues with some of the stereo tapes (why? more playbacks during the LP era to make laquers to replace worn out manufacturing parts?)
> Why 44.1/24 for the 2009 remasters? I have no idea except maybe there was a "beloved" but very old piece of hardware or software in their chain that wouldn't operate at 88.2 or 96kHz sampling rate?
> Regarding your point about super-hi-rez transfer to capture bias, doesn't that work only with your special heads and electronics? If you played a tape on a stock ATR-100 or Studer A-80 and captured at 192 or 384, would anything usable to Plangent be recovered?
> -- Tom Fine
> ----- Original Message ----- From: "Jamie Howarth" <[log in to unmask]>
> To: <[log in to unmask]>
> Sent: Friday, August 29, 2014 12:18 PM
> Subject: Re: [ARSCLIST] recording "cleanup" plugins and 192/24
>> It is essential to capture everything on the tape so that speed correction and distortion reduction can be performed. That's the fact of it - and it makes the discussion of audibility of 96 vs 192 moot.
>> You can't hear 100kHz or 152kHz (typical pre - ATR Ampex bias) but the results of capturing it and tracking it are extremely good, and beyond debate, at this point.
>> Once captured to 384 and processed do as you wish, do your declicking or editing
>> We also have a multiplex strategy for folding it to 24kHz for those who digitize direct at 96 that works well, but it requires our hardware.
>> Meanwhile however - drive storage space is cheap and at the higher rate through some heads and converters the time-base correction could be done whenever it was desired, often works from existing transfers played on 102s, which reach to about 120kHz stock, albeit at much lower S/n levels than ours. But still good enough to work wonders with the sound.
>> Working on a 1963 Abbey Road tape we found plenty of 80kHz raw signal when we transferred at 192/24 from an Abbey Road C37 Studer - pretty much their go-to machines (and J36) until 1967.
>> This confirmed that it would be relatively simple to dewow and deflutter every blessed Beatles recording from Love Me Do through Sergeant Pepper, using the existing digitizations at 192/24 if they existed.
>> But best as we know they don't, since Abbey Road standardized and defends 96/24. So until they pull the tapes again and do retransfers it's impossible, since the existing useful signal is lopped off. But to have that happen you have to get past the policy which is way harder than creating the technique. And get past this discussion into a "best practices" discussion about capturing the mechanical metadata of the recording. Which is easy.
>> Dan's paper as far as it goes makes sense, particularly back in the days when ADC settling time and clocking error and other flaws were less apparent at lower Fs... The converters worked simply did their job better at 96... No claims were made about utility or audibility, that's not his thing. From an engineering POV back then he was right... Pretty sure 192 and higher rate low bit over -sampling converters like the Pyramix 12.8mHz/5bit Horus (which sounds phenomenal) have caught up in the years since Dan wrote that paper.
>> Please pardon the misspellings and occassional insane word substitution I'm on an iPhone
>>> On Aug 29, 2014, at 11:33 AM, Lou Judson <[log in to unmask]> wrote:
>>> I appreciate your logic, but *I* feel upsampling does not give as much benefit as converting at the high rate. Delivering upsampled files claiming they are original transfers amounts to deception in my book. "… no one will know the difference" is the edge of a slippery slope ethically.
>>> Higher rates give better cleanup and processing for me, but I also feel 96k is good enough for anything... Agree with Lavry on that...
>>> Thanks for the article link!
>>> Lou Judson
>>> Intuitive Audio
>>>> On Aug 29, 2014, at 8:12 AM, Rob Poretti - Cube-Tec wrote:
>>>> If ABX listening is irrelevant from a business standpoint, then I would suggest simply up-sampling at the last delivery stage... no one will know the difference right? Why waste time and money? ... and you can still tout it as 24/192.