Any one using DAT machines?

[JCJR]

Reaper. Yay! Perfect!.Efficient and easy. I don't blah on about this DAW that much, but it is why I have it in my toolbox and why I will be buying the new version when 5 comes out.

But yeah, maybe Cool Edit is easier. That was a great program. I tried Audition but it was buggy.

Yep, have played with Reaper a bit and seems that it ought to be easy in Reaper.

I'm still running the final version of Cool Edit Pro before Adobe bought it. Still works great. The only issue maybe it doesn't write a compatible-enough 32 bit float wave file in that version. Had considered upgrading to Audition, but keep reading complaints that each new Audition version is no better than the final Cool Edit Pro. But never having used Audition, dunno one way or t'other.

[codec_spurt]

more of the same

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double post
mind fart

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ah go on

[JCJR]

While recording, are you monitoring off the thru of your computer sound, or monitoring off the analog audio out of the DAT?

If you are hearing clicks after the audio goes into your puter audio interface and thru the computer and back to the interface, you may be getting samplerate conversion glitches rather than DAT problems.

Just saying, if your DAT is sending 48K out the spdif, and your computer is actually recording some other samplerate, and the recorded files play back the same speed and pitch as what you hear from the DAT audio output-- In that case, SOMETHING either in your audio interface, or the driver software, or the host program (reaper) is doing real-time samplerate conversion or it couldn't be working.

If there are too many clicks to clean up with editing, later on clean the heads and try doing an analog dub to whatever sample rate and bit depth you want. Unless there is something bad with your interface or cables, a careful analog dub will sound as good or better than a digital dub with mis-matched samplerates.

[codec_spurt]

f**k it is

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Is it really this easy

[JCJR]

Sounds like you are having fun.

Apologies being a worrywart, but next time you get to the end of a tape and save your open files, try a little test if you want--

Unplug the spdif cable from the audio interface and verify that the interface and reaper are still set to your preferred sample rate. then play the computer file of a song, compare it to the same song on the DAT, and make sure they are both playing at the same pitch and the same speed.

The SPDIF sends word clock along with the digital audio. It could happen that WITH SOME COMBINATIONS of software and audio interface-- As long as the interface is receiving the 48 K word clock, the whole computer and interface will run at 48K even if you tell the sequencer program it is supposed to be working at 44.1K. So for instance you would be recording 48K into files labeled as 44.1K. And then when you remove the SPDIF cable, and the interface quits getting word clock, and it drops back to its 44.1 (or 96K, or whatever it was originally set to), in some cases you might notice that the 44.1K file that was actually recorded at slave 48K, will play back lower and slower, like a slowed down tape.

You have to remove the SPDIF connection, interrupt the word clock, before you can check and make sure this didn't happen. As long as the SPDIF is connected, then if the "wires are crossed" this way, then what you think is 44.1K will play back at 48K and it will sound fine.

Don't panic if that has happened. It should be fixable with samplerate conversion in software. Am just suggesting that you check before you get too far in, to make sure you are getting what you think you are getting.

[planetearth]

I see that I made most of my DATs at 48KHz too. Not a problem playing back on the machine, but I suppose it will be something to take into consideration when I digitally transfer to the DAW. The Panasonic just automatically adjusts, obviously, but this is something to be mindful of.

A sampling rate converter might be something for you.

But I realised I don't need not stinking sample rate conversion. If I record through my sp/diff into Reaper at 24 bits or 16 bits, or at 44.1KHz or 48KHz it doesn't matter does it coz it is being read in real time as an audio stream. NO encoding/decoding going on.

I think.

Hold on there, pardner!

You might want to check the manual for the Focusrite and your DAT deck to see what you need to transfer this stuff digitally. It seems you have everything you need for digital transfers; you should take advantage of this if you can.

Converting the audio from 14/15/16-bit digital back to analogue to send it over the cables and then convert it back to digital again (albeit at 24 bits) is not necessarily the best way to do this. If you can transfer the stuff digitally (which must be possible, given the gear you have) you'd get better transfers.

Steve

[Edit] Sorry, I just saw all the other posts that came in since I started my reply. You might have a handle on it all by now. Hope you do!

[codec_spurt]

major oblitoration sic

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just testing here to see what is possible

[Compyfox]

I see that I made most of my DATs at 48KHz too. Not a problem playing back on the machine, but I suppose it will be something to take into consideration when I digitally transfer to the DAW. The Panasonic just automatically adjusts, obviously, but this is something to be mindful of.

A sampling rate converter might be something for you.

But I realised I don't need not stinking sample rate conversion. If I record through my sp/diff into Reaper at 24 bits or 16 bits, or at 44.1KHz or 48KHz it doesn't matter does it coz it is being read in real time as an audio stream. NO encoding/decoding going on.

I think.

Hold on there, pardner!

You might want to check the manual for the Focusrite and your DAT deck to see what you need to transfer this stuff digitally. It seems you have everything you need for digital transfers; you should take advantage of this if you can.

Converting the audio from 14/15/16-bit digital back to analogue to send it over the cables and then convert it back to digital again (albeit at 24 bits) is not necessarily the best way to do this. If you can transfer the stuff digitally (which must be possible, given the gear you have) you'd get better transfers.

Steve

[Edit] Sorry, I just saw all the other posts that came in since I started my reply. You might have a handle on it all by now. Hope you do!

Yeah, I also lost track of what's going on. But... since I've done this a couple of times by now:
Digital transfer is LOCKED(!) to the sampling rate that the source if sending.

Example:
If DAT is setup to 44kHz, you can only (ideally) transfer to 44kHz at either bitrate through the SPDIF pipe. The ADC basically syncs with the DAT machine. If the SRC's are off, either no sound to record, or a pitched one.

UNLESS... you use something like the Behringer Ultramatch. Here the source and the destination can be different. And this device is clearly made for that purpose.


In my example:
If I'd copy digitally from DAT to HDD without SRC, then I am forced to use 44kHz. But since i can set up the device to my needs, I can record in 96kHz in Wavelab/Cubase while the DAT plays back at 44kHz. With or without the copyright flag being active - doesn't matter, I can copy in the digital realm to my hearts content.


Now if I'd go the analog route, I don't need to worry about SRC; but rather noise and the likes. So digital transfer is indeed the preferred route.

[JCJR]

Converting the audio from 14/15/16-bit digital back to analogue to send it over the cables and then convert it back to digital again (albeit at 24 bits) is not necessarily the best way to do this. If you can transfer the stuff digitally (which must be possible, given the gear you have) you'd get better transfers.

My rationale for an analog dub, if too many problems are encountered-- When PCM/Video recorders, and later DAT first came out, recording engineers experimented doing ridiculous generations of analog dubs between the (then newfangled) digital recorders, discovering that many analog back-and-forth dubs were necessary before noticing audible degradation. Of course, such fellers were presumably real good doing analog dubs, having had lots of practice.

Ergo the theory-- ASSUMING a modern puter interface is at least as good as were the first generation of digital recorders, then the ear would not likely hear a difference between a single-generation analog dub, versus a digital dub.

But if the digital dubbing is working fine, then that is better because it would eliminate any doubt about hum in cables, analog noise, possible inadequacies of some random selected modern audio interface, proper level-setting to avoid clipping, etc.

[ontrackp]

I still have 2 SV-3800's, a SONY DAT deck and a TASCAM DAT machine from the good old days. When DAT first came out we would run 1/4" tape in the background as a safety net in case the DAT tapes failed.

It was not uncommon for tapes recorded on SONY machines to not work on Panasonic and vice versa, so many studios had machines from each manufacturer.

The best thing I ever heard about DAT was from an engineer friend of mine many years ago, during the 5 years or so that DAT was the current tape format, that was:
"It's a miracle it works at all".

One of the best things to happen in the world of music production was that DAT went away.

[planetearth]

Converting the audio from 14/15/16-bit digital back to analogue to send it over the cables and then convert it back to digital again (albeit at 24 bits) is not necessarily the best way to do this. If you can transfer the stuff digitally (which must be possible, given the gear you have) you'd get better transfers.

My rationale for an analog dub, if too many problems are encountered-- When PCM/Video recorders, and later DAT first came out, recording engineers experimented doing ridiculous generations of analog dubs between the (then newfangled) digital recorders, discovering that many analog back-and-forth dubs were necessary before noticing audible degradation. Of course, such fellers were presumably real good doing analog dubs, having had lots of practice.

Ergo the theory-- ASSUMING a modern puter interface is at least as good as were the first generation of digital recorders, then the ear would not likely hear a difference between a single-generation analog dub, versus a digital dub.

But if the digital dubbing is working fine, then that is better because it would eliminate any doubt about hum in cables, analog noise, possible inadequacies of some random selected modern audio interface, proper level-setting to avoid clipping, etc.

True, I doubt anyone using KVR would hear the difference between first- and second-generation copies of his DATs, even if the second-generation was an analog transfer. But he seemed to be very concerned about the audio quality and the signal-to-noise ratio, so I thought I'd just mention it.

Besides, an analog copy is better than no copy at all!

Steve