Your opinion, tape mastering of digital music

Hi,

I've been recording music on an ancient Apple G5, a Presonus DAC amp, and Cubase software. I'm happy with the playback of mixes I've made, but when my computer does a mixdown the results can sometimes be disheartening. In recent years a friend of mine gave me a Teac 3340S in great condition. I've considered recording the computer playback into the Teac 15ips and later recording that tape playback into the computer. I'm hoping Teac would better capture my mix and possibly add an aesthetic enhancement. I'm a complete novice, but I know there's studios out there that offer tape and analogue treatments for digital content. I'm curious to hear from anyone who may have experience with this sort of thing. One question off the top of my head is, when I run the playback from the computer into the Teac, should I do so at a lower level and boost it within the tape machine, or should I run it in moderate to strong? I also heard that in analogue there's more forgiveness when the vu meters clip in the red; obviously I don't want to amplify it until it becomes undynamic--so what should I look for? I've ordered some 1/4 inch tape and will be experimenting, but I'm curious what y'all have to say and if you could give me some pointers so I don't start out too stupid.

 

What version of the Mac OS are you running on that G5, if you don't mind me asking?

 

10.4 which is ancient. I plan on getting a new computer and software next year, but at the moment I'm in too deep with a project that I'm eager to mix and dress it up to get it out early next year. I love recording and arranging, but this side of the equation I've never gotten into and I hope I can give it a go.

 

That's impressive, maybe I should stop updating my Mac.

 

I think any enhancement you would be adding would be called tape hiss. If that's the aesthetic you're after, go for it. But I wouldn't do it. If I would going to try anything, I would run the sound out of a DAC into a Tube Buffer then back into an ADC to get it back to digital. You could play with different tubes to see how it affects the sound. But going through all the trouble to mess with a R2R deck just seems like a pain. You'd have to record it. Then rewind the tape followed by running the tape back to the take-up reel. There's a reason R2R never really captured the general public. But somehow the cassette tape did. The reason is simple, you don't have to mess with that take up reel. Again if you're after tape hiss, you could give it a go. But, I'd look at a digital solution first.

 

Hrm. This is not a Studer or Otari from the 80's. It's 1972. You'll be able to mimic the sound of bedroom 4-track recording.
Outboard Dolby unit available, (2 channel AN-180 on eBay right now) if you want to get this close to a metal cassette played on a 1995 JVC.

This deck can be used like a 3-head cassette deck: the output can be set to the playback head while it records. For that matter, you could record on three seconds of spliced tape loop just long enough to slip around the reels.

Multitrack software is able to playback while recording to other tracks simultaneously. You could punch in one stereo channel through all four tracks mono on the R2R deck for the lowest noise, then repeat for the other side.

With tape, the effect and tone you're looking for is the tape saturation & hysterisis. 15 IPS doesn't self-bias as much from loud signals, so you'll be able to crank it up; in fact, you'll need to in order to overcome -55dB noise.

 

lol, much rather be in your boat at the moment.

Thanks for advice. Neat idea on the Dolby unit.

Bit confused, are you saying that I should record to tape one stereo channel at a time?

 

I really appreciate the advice I've been given. The main problem with my mixdowns is that my computer will do a vanishing act with my piano tracks and guitar solos. I'm hoping I can find a way to record my computer playback on tape so that I don't have to ask my computer to do mixdowns. If that don't work out, I'll just do what I usually do--make purposefully wonky mixes where keyboards and guitars have to be mixed super hot in order to have the presence I'm looking for after a mix down.

 

Okay that makes sense to me now. Reduce tape noise by occupying all four tracks.

That Teac AN180 noise reduction machine is pretty expensive. I noticed a few other Teac Dolby NR machines at lower cost, most of which were ones from the 70's. Is that An180 really a game changer compared to others?

 

It sounds like something could be off with your settings in Cubase. There shouldn’t any difference on what you hear on playback and what you hear on mixdown. A couple guesses; are you mixing down to mono? Do you have any busses or subgroups that aren’t assigned to the master on mixdown? That could be causing the issues with certain elements disappearing from the mixdown.

I use Logic and not Cubase, but you should be able to try something like this. Assign all tracks and busses to a new bus, effectively creating a new master bus. Route this newly created bus to a new stereo audio track. Record your playback to this new audio track (route to the master and not to the new bus you created that you assigned everything else to) and it should be exactly the same as what you’re hearing on playback.

 

I regularly run digital to tape in my home studio and the other way around. The advantage you might be looking for is called tape saturation and will provide a certain "natural" sound that it the result of tape EQ, midrange boost, and dynamic compression. In general, the higher level you record at, the more you get this tape "sound."
My tape equipment is all balanced and Otari decks that run at +4 dB. Consequently, you don't want to record these much past 0 dB on the meter or you will explode the rest of the analog chain. Your Teac is different, but check to see whether it is +4 or -10 output.
Have been a mostly digital guy for 35 years and was surprised to find out that analog tape could be so good and that moving feeds back and forth from digital could offer such pleasant listening results. Best to try rather than build on digital theory.
Am using a Dolby 363 unit with Dolby SR to get the maximum in analog recording at 30ips on an Otari MTR-10. It sounds so good I have been stunned when comparing it to a RME UFX 192khz, 32 bit digital recording.

 

I'm recording stereo at 24/96 and mixdown the same.

I usually associate busses in recording as correlating connections between mic/instrument to recording tracks in software. So I'll need to look into what you suggested to understand what to do. I think you may be on to something.

 

Feel free to PM me some screenshots of your mixdown options and/or mixer window if you aren’t able to find the issue. I’ve done something similar in the past where I had a reverb aux send that wasn’t routed to the master, but the way my session was setup I could hear it on playback. It was driving me crazy that the mixdown didn’t have any reverb and it was a very simple fix once I double checked all the routing.

 

I looked into all the routing and everything looked right. But, while checking it out I found something that stood out. For years I've had my software preset to record at 24/96--and I'll be damned if it wasn't set at 32/96. If you asked me earlier today I would have told you that I've never worked with 32 bit, but I guess I've been doing just that for I don't know how long.

My theory now is that certain instruments have been losing presence because I'm doing 24/96 mixdowns of a 32 bit playback.

Now, AudioAddict you work regularly with 32 bit--does this make sense? Even though they're both at 96, would you really lose significant presence of instruments between 32 and 24?

 

Moving between 24 and 32 bit this way should not present much of a difference -- depending upon your complete audio chain. My DAW, Samplitude, uses a floating bit rate that can exceed 32 bits but then you choose a value when you mix or export. 32 to 24 should not make much of an audible change in presence. Many studios use 24 bit as a standard and 96/24 has been used widely for 25 years. Have been using 192/32 in recent recordings and find a very small audible improvement that is mostly discernible on my best digital studio monitors (Genelec 8351s). Once a mixed digital file is created, then there is little audible change between 32 and 24.
Having said that, would like to suggest that a more important quality factor is maintaining the *.wav format. People talk about lossless compression, flac formats, and other new dodges (MQA as an example) but I have found that trying to save money by minimizing memory is futile. Memory is relatively cheap by audio standards -- a 4tb drive can be as little as $150 (and roughly $450 for a SSD). You can put a very large library on this or record a lot -- even at 192/32.
But when you switch to another file format, such as Wave to Flac, then you transform the waveform and cannot be certain that it will be recreated perfectly. Keep everything *.wav; you will hear one studio person after another preaching this. Output to a *.wav format and, then, should you export to tape, you will achieve the best waveform accuracy that you can with the slicing and slithering that goes on with digital.
Have been recording simultaneously with both tape and digital at the highest levels -- tape at 30ips, 2-track, and digital at 192/32, then comparing the two by switching between the sources. They are VERY different. Each has its advantages but for sheer listening pleasure, I always select the tape source. As a musician, the tape sound is clearly more accurate for waveform integrity and microdynamics; acoustic instruments sound natural and "correct." Digital sounds quiet, beautifully centered in soundscape, and has a facade of extreme accuracy. But the more you listen to digital, the more you notice that the instrumental timbre is "flat." The subtle shades of timbral shifting that occur over time are not there.

 

Well said regardng wav and the difference analog makes with timbre.

 

Thanks so much! Very interesting. Question, I've read elsewhere that when one works with higher tape speeds, the tape noise isn't that noticeable and that you can touch it up later with eq to reduce hiss. What's your take on Noise Reduction? Does something like Dolby type A work better with reel tape?

 

The last album I completed was tracked/overdubbed/mixed in Pro Tools, then we printed to a Studer 1/2" before playing it once again back into Pro Tools. No NR. I really liked what tape did to most of the songs. That light bit of tape compression really was the glue that made the mixes hang together. You will find that some results are better than others, that not every track will benefit much or seemingly at all, in some cases.



Dan

 

Yes to both questions; tape hiss is very slight, but still audible, at 30ips and each step up the amount is lessened. However, 15ips has a good amount of hiss (don't believe otherwise) and requires Dolby to get the sound digital-like. While you can use EQ to remove hiss, you will also remove WAY TOO MUCH musical content, so I avoid this step. If you have to do this, use an analog notch filter such as the Art EQ355.
Have been comparing Dolby A and SR with a Dolby 363 unit and both are good. "A" is the older standard the industry used until SR became the norm in the 90s. You can still hear a small amount of "pumping" with A if you listen carefully in a quality studio setup. This is because the noise level frequency adjustments are fixed. SR is a whole 'nother level -- it is Dolby's magnum opus and does the best that noise reduction can do. Movie studios are continuing to use it.
Dolby SR on a 30ips deck that is correctly aligned and calibrated will startle you with its accuracy -- even if you are used to world-class digital.
If you are looking for a system for your Teac, either A or SR will work but remember that the Dolby 363 unit requires a mixer on the inbound end. Both the deck and mixer sides have to be calibrated carefully. Have made a "remote" for my unit that automatically changes the 363 from play to record based on the deck's operation. This is very handy.
BTW if anyone has got a Dolby 363 unit, PM me -- I will pay top dollar for another unit.

 

Thank you everybody, so much is clearer to me now. Really enjoyed reading your input and experience.

I think my mixdowns are a bit much for a computer my age and mileage. At the moment I'm editing drum tracks to avoid using gate filter and eqing other tracks individually to prepare for a final mix. When I get there I'll make a judgement call; if mixdowns are still a bit shifty then I'll look into a NR and give tape a go.

Feel free to hit me up with more advice if anything else comes to mind. I'll report back later after making more progress. Thanks.

 

I’d consider recording on the Teac and then importing those tracks in to Cubase for mixing/mastering. But even then the Teac might need some restoration, so I think you’d be better served using that $ for a new Mac. The new Mac Mini’s look very tempting to me for home recording.

There’s some plug-ins that attempt to recreate a tape source that might be worth looking in to as well.

The Teac is 1/4”, right? If so, dividing that up in to 4 tracks gives you only 1/16” so the fidelity will be compromised regardless of any other factors.

 

Very unlikely.
Most audio software uses 32 (float) because whatever you do in digital post processing, it is a calculation and each calculation has a roundoff error (better known as quantisation error).
Try storing 1/3 in an integer

To avoid this,the project should be 32 float and not 24 integer.
Only the finalised product should be exported to 24 bit.

Observe the loudest a DAC can play is 0 dBFS (all bits on to put it simple)
A 24 bit recording goes down to -144 dBFS.
Even if you have some some very, very quiet gear you will struggle to reproduce anything below -120 dBFS
Obvious your problem is not the bit depth.

 

Print it to tape as strong as possible without saturating it if you are using 15 i.p.s.. You'll want to fight tape hiss and preserve dynamics and transients.

But, the main thing is to experiment. Maybe you'll find that you prefer the digital sound after all. Who knows?

 

Dolby 363 Stereo SR Noise Reduction Unit

 

Now that's something I could definitely get behind in my next project. At least like what Grant says and try to just to see if it works.

'And you know that notion just crossed my mind'... Well, all I can say is that the R2R sounds great and the only tape it came with, "Sam The Sham & The Pharaoh's Greatest Hits" sounds stupid good on it. I hear harmonics in the electric combo organ that makes it sound way more futuristic than frat rock should. But you are correct--that is a finite little space to put all that data into and it is a bit concerning.

Totally, that Mac Mini is very nice and seems perfect for audio. Those big Mac towers are really meant for video editing and hurting your wallet. The Teac fortunately didn't need much restoration. The previous owner collected it as collateral from a tenant back in the 70's. He never used it and decades later gave it to me. I took it to a sound tech a couple years ago and all he had to do was minor maintenance like changing out the rubber parts. The tech told me that the tape heads are like new. Sometimes I listen to hi res digital music through the Teac like a pre-amp and it gives the music that punchy sweet analogue sound. Everyone from Kris Kristofferson to Beyonce sounds great through that Teac. It's a fun toy in that regard, haven't used it for much else.

I can relate--I was in a long distance relationship once.