Panasonic SV-3700 to Pro tools?

G

ginridge

New member
I have a series of recordings from the 1990's that were mixed to DAT on a Panasonic SV-3700 that I need to put into pro tools. I've located an operating SV-3700 and need figure out a way get those tracks into pro tools 12. I have a recent Apogee Ensemble Thunderbolt interface. I also have an older Mackie Digital 8 buss that I don't really use anymore, but might have connection that can be of use.

The Panasonic has AES/IEC and IEC Type II digital outputs, the Apogee Ensemble has S/PDIF and Thunderbolt ins/outs.

Is there a way to interface these units?

I'm wondering about going out of the Panasonic with AES/IEC into the Mackie, the out of the Mackie via S/PDIF into the Apogee via S/PDIF.

I'd appreciate any suggestions or advice. Thanks!
 
You 'should' be able to use the spdf and go digital to digital.

I've dumped a lot of my stuff from dat to PT.

But I'm using a Tascam and a Focusrite interface. Both have spdf.

Why i said 'should', is because when I used spdf, it went into PT too hot. I don't know why. Maybe there's something I could have done in protools, but I wasn't that familiar with it yet.
So I just went analog out of the dat into line in on the interface. It worked fine and I never looked back.

But anyway, try your spdf first. Less conversion involved. I'm not familiar with the mackie.

Worst case go out of the analog output of the Panasonic into the apogee. On my dumping I can't tell ANY degradation because of the additional conversion.

I would recommend to transfer as 2 mono tracks (left and right) rather than one stereo track.
:D
 
Why would a clock have to be set?
Click to expand...
One or the other always has to be the master, the other follow that for it's clock source ('slave.
If the DAT has a clock input, it could take that as it's sync source from the I/F's clock out, I/F remaining the master.
 
RFR said:
Not familiar with those connections. Why would a clock have to be set?
Click to expand...

They're both SPDIF.

The receiving device needs to sync its clock to the incoming signal so the digital data can be read correctly. Theoretically the receiving device could send sync signal to the send device, or you could use an external clock in some cases, but with SPDIF it's generally simpler to sync to the incoming signal.
 
Thanks for the clarification. What confused me was this. When i hooked up my spdf frpm my Tascam DA 30 to my focusrite saffire pro40, I only hooked up the spdf cable. No setting a clock was involved.
As stated earlier, only thing non optimun wss it ended uo too hot in protools.

Admittedly I'm not a tech genius. Just an old analog hack. :D

Anyway, sorry for the thread hijack.
 
There is a voltage mismatch between the ebu balanced output and the spdif input but this doesn't lead to any level changes in the audio files, so if it's hot in protools it was hot in the track. In the early days of digital we just did what we did in analogue. As hot on the peaks as we could without going over, and often we did! Unbalancing the xlr and going in via the phono connector normally works. As for the clock, if your system doesn't mind drift, it will lock, but many are quite sensitive to it, so setting the input to take the clock from the external source is sensible. It either works or it doesn't, so whatever is stable is the one to use.
 
I'd be curious why (or how) you got 'too hot of an input on a digi transfer as well.
Might it have been the sources were hitting right up at 0dBFS with the Saffire interpreting them as 'overs'?
Thanks.
 
rob aylestone said:
In the early days of digital we just did what we did in analogue. As hot on the peaks as we could without going over, and often we did!
Click to expand...

[MENTION=7442]mixsit[/MENTION], I think the above quote contains the answer.

In those early days, that was the attitude, at least for me. All I knew about digital was that going over zero sounded nasty, unlike tape where it was acceptable.

So you pushed the inpur level as high as you could with no "overs". Some compression on the 2 buss was used, but a shitty one.

A limiter??? What was that? Oh, something only rich people had, or the Record Plant.

Mastering? Again, only rich people, or those with a record deal.

The point is, the goal was to get it to sound like a record coming off the board with limited resources.

This makes me wonder as to the accuracy of the DA30 metering.

I have a Durrough digital meter. (Hardware not plug in)

I think I'll hook it up to the Dat machine and see what my actual levels are.
:D
Hooking up the RCA outputs to the focusrite inputs and dropping the dat output levels did the trick.


I sure hope this thread is of some use to the OP with it's twists and turns. :D
 
I used to get loads of performance playback tracks on MD - loads went to full on the meter and stayed there for 3 minutes - oddly, the ATRAC compression used didn't sound distorted and nasty - so you just got this LOUD weird sounding compressed thing. So many people over recorded on mini disc because it seemed to cope, whereas DAT sounded dreadful.
 
Excellent responses! Thanks very much. Let me see if I have this correct, I should be able to go out of the DAT's IEC Type II output (RCA) and go into the Apogee's SPDIF (also RCA). I just took a look and the SV-3700 does NOT have a word clock cable out. Will that be a problem?
 
ginridge said:
Excellent responses! Thanks very much. Let me see if I have this correct, I should be able to go out of the DAT's IEC Type II output (RCA) and go into the Apogee's SPDIF (also RCA). I just took a look and the SV-3700 does NOT have a word clock cable out. Will that be a problem?
Click to expand...

The Apogee should receive its clock from the SPDIF signal itself. If it doesn't automatically sync up there may be a manual option in the Apogee control panel software.
 
bouldersoundguy said:
The Apogee should receive its clock from the SPDIF signal itself. If it doesn't automatically sync up there may be a manual option in the Apogee control panel software.
Click to expand...

So along that logic, it's possible that my Focusrite would automatically sync up with the Tascam and recieve it's clock from the spdf of the DA30?
 
Thanks again for the clarification. I spent about an hour on the phone w/ Apogee support and they have shown me what I'll need to do to make this work. I had to into the Apogees software program, Maestro, and re-set the inputs and a few other minor adjustments. Unfortunately, I do not have the coax cables (just ordered them, should have them next week) so I can't tell if this will do the trick, but it looks like it should. I'll let you know how it all turns out.
 
I should have mentioned this before, but I assumed you had a cable. Video grade cables with RCA ends work for SPDIF, so all those component and composite cables that have been made obsolete by newer TV technology are fair game. If you want to try the setup before your official SPDIF cable arrives, grab a video cable.
 
I think we should set the record straight a little on the connections. AES/EBU is NOT exactly the same as SP/DIF. Electrically, the balanced XLER output on the 3700 needs a 6dB pad. 2.7V out of the Panasonic, into an input expecting .5V. Some people report no issues, other report no success. I've not seen any reports of the voltage causing damage, but you should really reduce the level to do it properly. Second, the actual data is similar, but different too. There is a switched but in the status frame - SPDIF uses it as a status bit to prevent consumers doing digital copies - the 3700 doesn't send it or pay attention to it on the input. SPDIF by design is self-clocking, taking sync from the data coming in. The SPDIF is a 75 Ohm impedance connection and Long runs don't work as cable capacitance ruins the incoming data and at some point it stops. Modern SPDIF receive chips are much more tolerant of dodgy data - so the system should work. Back in the days these machines were current, connections often failed. Now, they're better.

You may find it works with just the cable, however, if it doesn't the problem is likely to be too much signal, and you need to make a pad - which will also handle the balanced to unbalanced conversion.

Here's a circuit for the pad

Screen Shot 2018-02-24 at 17.54.17.png
 
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