The ATR 28.8 kHz buzz is traceable so that’s not bleak.
We can pull Studer 240kHz easily at 30 a little tougher 15, and at 7.5 not yet. Due to a fluke in the PCB layout 80kHz erase is pretty strong from the record head, and that’s easy.
As for the later 3M and MCI in the ca 220kHz the system has a multiplexer that translates any bias that comes at it to 24kHz. So it’s easy to archive at 96/24. Been that way since 2003.
As John pointed out 384k Fs catches almost every machine directly.
I’m glad Richard has picked up on the IM in the Dolby, I had talked to GML about that quite a while back. The predominant quality improvement with flutter remediation is a dramatic drop in IM. The Dolby issue is real but it’s down in the mud compared with the interference caused by the transports.
Please pardon the mispellings and occassional insane word substitution I'm on an iPhone
> On Jul 14, 2019, at 20:53, Richard L. Hess <[log in to unmask]> wrote:
> Hi, Corey and Gary,
> Thanks for your kind remarks about the decoder. My colleague and friend, John Dyson has done a wonderful job with the code. His acid tests have been leaked Dolby recordings of 70s pop music--some of them sound so bad until he decodes them...but they are tougher than the stuff I've recorded and obtained from other sources.
> What has happened is the intermod that is normally generated by fast gain changes on decoding is vastly reduced.
> As to my question, am I the only proponent of recording the raw, undecoded output? It's saved my bacon more than once, and I've been insisting on it for at least a decade. I was hoping that some standards/best practice body recommended it. I did not think I was alone.
> John Chester, thanks for the info on 384 kHz sampling frequency and bias.
> Remember my effort here?
> The only major recorders that are problematic (i.e. bias frequencies above 180 kHz are:
> Ampex ATR-100 (432 kHz)
> Sony APR-5000 and probably multitracks (400 kHz)
> Studer A80VU (240 kHz, most late models are 150 or 153.6 kHz,
> the A77 is 120 kHz)
> Otari MTR-10/12 and MTR-90 (246-250 kHz)
>> On 2019-07-14 7:16 p.m., Gary A. Galo wrote:
>> Hi Richard,
>> I echo Corey Bailey's email in congratulating you on the software-based NR decoder. I'm sure there will be a considerable market for it.
>> The issue of preserving the "original" data - whether analog of digital - is a sticky and controversial one. When I gave my ARSC presentation on transferring PCM-F1 format digital recordings for the NY ARSC chapter April 2018, I was taken to task by one attendee for not preserving the original bits. I go from the S/PDIF output of my PCM-601ESD digital processor directly into a Tascam DA-3000 digital recorder. The Tascam has a built-in, switchable sample rate converter based on the Cirrus Logic CS8422 SRC chip (which doubles as the S/PDIF input receiver). I set the Tascam to record at 88.2 kHz, so the CS8422 is converting 44.056 to 88.2. An "undocumented feature" of the DA-3000 recorder is that the CS8422 SRC chip also does 50/15 uSec de-emphasis, which take care of another issue with F1 recordings. Why Tascam fails to mention this anywhere in their manual or product literature is beyond me, because the de-emphasis feature is clearly stated on the front page of the CS-8422 data sheet, and it's an extremely useful feature.
>> With this method, only the inter-channel time delay and DC offset still need to be addressed once the 88.2 kHz data is on your computer.
>> My method does not save the original 44.056 kHz bits. Guilty as charged. But, the CS8422 does a beautiful job with the SRC and the de-emphasis, and has ultra-low jitter clock recovery to boot, so I sleep well at night. If you feel the need to preserve the original bits, you could run a second, raw transfer directly into your computer, if your computer will lock onto 44.056 kHz. Or, you could use a digital distribution device to split the 44.056 kHz data stream, sending it to both the computer, and the DA-3000 recorder simultaneously. But, I just don't see the need.
>> So there is no misunderstanding, I can well understand the desire to preserve the non-decoded Dolby-A analog signal in case better software conversion becomes available down the road. It makes sense to do this. So, perhaps I'm being inconsistent. These are thorny issues, and everyone will have their own viewpoints.
>> Gary Galo
>> Audio Engineer Emeritus
>> The Crane School of Music
>> SUNY at Potsdam, NY 13676
>> "Great art presupposes the alert mind of the educated listener."
>> Arnold Schoenberg
>> "A true artist doesn't want to be admired, he wants to be believed."
>> Igor Markevitch
>> "If you design an audio system based on the premise that nothing is audible,
>> on that system nothing will be audible."
>> G. Galo
>> -----Original Message-----
>> From: Association for Recorded Sound Discussion List [mailto:[log in to unmask]] On Behalf Of Richard L. Hess
>> Sent: Sunday, July 14, 2019 5:42 PM
>> To: [log in to unmask]
>> Subject: [EXTERNAL] [ARSCLIST] Preserving both raw and decoded files for tapes recorded with Noise Reduction?
>> Hi, I think many of us agree that it's necessary to preserve both the
>> raw transfer and the decoded version of a file which has been recorded
>> with Dolby or DBX type noise reduction.
>> When I first thought about it, I never imagined I'd be part of a team
>> that would produce a better decoder for Dolby A encoded tapes than
>> Dolby, but it's happening and humbling... So, it is a good idea to save
>> as much raw data as possible because who knows what else will come along.
>> Plangent is wonderful, but a bit problematic as it is still inconvenient
>> to properly archive the bias, but that's another story, and I think in
>> the long run it would be good if we could do that.
>> MY QUESTION is: Are there any standards or recommendations that say
>> "keep the raw undecoded copy as well as keeping the decoded copy?
>> It's for a paper that Federica and I are writing.
> Richard L. Hess email: [log in to unmask]
> Aurora, Ontario, Canada 647 479 2800
> Quality tape transfers -- even from hard-to-play tapes.