Your post raises many questions.
First off, the vast majority of DATs were 16-bit devices running at 32,
44.1, and 48 ks/s. I say "vast majority" because Sony had a 12-bit
"Long-Play" version (which the DA-20MKII can also reproduce, but
Panasonic machines cannot) and also because I do not know what various
close-to-end-of-life iterations were made in the format for "HR" or high
resolution recording. I think there were a few 96 ks/s machines
available but I do not know whether these were capable of higher bit
So, the first question is are the DATs outputting 16 bits while the
software is locked to expecting 24?
Next (or perhaps first) I would worry about the clock settings on the
interface and the software. Not only do you need the clocks set the
same, you need to have the interface locked to the incoming digital data
or what you describe will happen. I do not know if there is any way of
locking the DA-20MKII to a clock source. I always lock my interface to it.
In my studio, I use RME interfaces (two Multiface II units and a
Fireface UFX). All of these nicely fail over to internal clock when the
DAT is not running, but when I used a MOTU 828 MK II this summer to do a
mass ingest of DATs at a separate workstation, I found I had to have the
DAT on and at the correct sample rate in order to play back the file
since the MOTU fails rather than fails over to internal.
If all this is working, activate the error count on the Tascam (it's a
nice four-digit counter explained in the manual or supplement available
from the Tascam website). Quick Notes: error rates are 1/10,000. Press
COUNTER MODE while holding the REC MUTE while in Stop or Play. Pressing
play starts the count, but the display will be dashes for the first few
seconds until the rate can be calculated. AUTO ID cycles the display
through AHead, BHead, ABheads.
If that error rate is more than 0100 or so, I would suspect the
DA-20MKII is concealing errors in its audio output that it is not
concealing in the digital output if all the clock issues are addressed.
This is SPECULATION on my part, as I do not know the precise flow. Also,
if the tape is pre-emphasized, I do not think that the de-emphasis is
provided on the digital output (that could make it sound noisier).
Samplitude/Sequoia's FFT filter has a de-emphasis filter preset that
seems to work according to my friend Bill Lund who has more reason to
use it than I at the moment.
Anyway, there is an explanation for this...but I'm not certain I've
covered all the possibilities here.
On 2012-08-28 9:49 AM, Brian Carpenter wrote:
> I couldn't find this question addressed in the ARSCLIST archive, so,
> apologies for any redundancy. I'm transferring a set of DATs, recorded at
> 48/24, using a TASCAM DA-20MKII and WaveLab 6 software. The .wav files I
> end up with in WaveLab are highly, highly glitchy. Errors and dropouts
> galore. However, when I listen to one of these tape directly from the deck
> itself, it sounds completely fine. No problems at all. I've transferred
> hundreds of DATs and have never encountered this before. Any ideas as to
> what's going on here?
> Any pointers much appreciated!
Richard L. Hess email: [log in to unmask]
Aurora, Ontario, Canada (905) 713 6733 1-877-TAPE-FIX
Quality tape transfers -- even from hard-to-play tapes.