The Saffire Pro 40 has 8 mic/line inputs, 8 channels via ADAT lightpipe and 2 channels via S/PDIF on RCA sockets. I haven't seen a detailed block diagram of the unit, but my assumption is that, in common with most interfaces of this type, the analog line inputs are attenuated and put through the same pre-amps as the microphone inputs. That would mean that using a high-quality external pre-amp feeding the Pro 40 line inputs would still give you tracks that have the Pro 40 sonic signature.

If you want to avoid this, you have the option of two forms of digital input: 8-channel ADAT on a lightpipe or 2-channel S/PDIF on an RCA (phono) connector. Audio sent through either of these digital forms is not modified in any way in the Pro 40 and so retains the character of whatever sent it. However, all three of the external pre-amps you listed are (as far as I know) not available with internal converters and hence digital outputs, so their analog outputs would either have to be fed into the Pro 40 line inputs or you would have to add yet another piece of gear - a line-level ADC unit.

If you were prepared to consider other high-end pre-amps and channel strips, and specifically those that had integral ADCs, you might want to look at something like the API A2D, UA 4-710D or AMS Neve 1073 DPD. These all have either ADAT or S/PDIF connectivity and so would connect to the digital inputs of the Pro 40.

Another route to consider is the one you mentioned about trading the Pro 40 for another interface that would be more suited to the direction you want to go down. The interface unit that comes to mind here is the RME UFX, since the line inputs on the RME range do not go through mic pre-amps, resulting in remarkably clean conversion. One or more of your chosen analog-output channel strips would give good results through an interface of this type that allows the quality of external pre-amps to shine through.

need more info on the pres in this unit.

Originally Posted by geerhakerz.com/salez_bs

Focusrite's unparalleled preamp legacy forms the foundation of this new interface, with eight award-winning Focusrite preamps ...Class A Pre Amp
OctoPre
This pre-amp demonstrates the same
wide-bandwidth philosophy featured in the
original Focusrite units from years past.
Focusrite’s custom pre-amp ensures low noise
and distortion, delivering clarity and warmth
without unwanted artefacts, displaying the
signature sound for which Focusrite has
become famous. Also found in this section are
Phantom Power, a Line level switch and a high
pass filter, enabling OctoPre to handle a
multitude of audio sources simultaneously.

nothing on what type of pre it is. if it's a decent design, good parts / chip ... maybe it would sound fine trimmed down for line inputs. i would guess its a glorified fixed gain type electronically balanced with trim on the output stage. ... the specifications say; "Automatic switching of Mic / Line (e.g. inserting a jack switches from Mic to either Line or Instrument" ... what the heck you have it, go ahead and try it. might work just fine.

using adat will limit sample rates to 48k

Here is the rub with these types of computer audio interface devices. Many of the line level inputs are simply unbalanced wiring variations, still generally going through these pre-amplification circuits. These pre-amplification circuits used in this way and capacity, pretty much just makes them, low color, simple buffer amplifiers. Which isn't much different than the interior wiring and concept design, inside many audio consoles. So it's certainly not a horrible way to go at all.

Yet with these types of outboard preamps, there headroom and output drive capabilities, generally can deliver output signals, that will exceed the input overload capabilities, of those computer audio interface devices. So while you will still obtain that quality tone, you'll certainly not want to overload the inputs of the computer audio interfaces. And where you will find some of these outboard preamps offer output level controls that are actually passive, on the preamp outputs. And this serves two purposes. One is to prevent overloading that input stage of that combo microphone/instrument/quasi-line level input. And so one can also take advantage of the gorgeous sounding overdrive of the electronics from the preamplifier, saturating the output transformer, of the preamplifier. Which means over driving the output circuitry until it starts to get a little nonlinear and raggedy while it also saturates the transformer on the output providing a smoother feel of the additionally generated harmonic distortion components. And this is where a lot of the magic with those outboard preamps, happens. Frequently negating the need for dynamic range processing and even equalization. Yeah baby! So those preamplifiers give you a broad range of generating different tonalities and character, far beyond what that relatively neutral sounding computer audio interface, features.

Of course those very nice Lavery converters, costing upwards of $3000, with only two input channels can be breathtaking sounding. And which you might find to be of a more critical requirement for Operatic/Symphonic and other highly esoteric oriented productions. But when it comes to rock 'n roll... a lot of rock 'n roll is based upon the sonic artifacts of different types of harmonically perceived distortion elements. And where I think, reasonable average, IC chip, analog to digital converters, can offer up sonic quality, far beyond what similar equipment could just a few years ago. I mean how much perfection does one need in an imperfect world? It's only rock 'n roll and I like it, like it, yes I do, anyway I can get it. I mean I don't even expect to hear rock 'n roll without some kind of distortion playing a big part? And so if someone's analog to digital device begins to Peter out around +18/+20, it sure as heck won't ruin my recording. Even though I might in fact be missing out on 10 DB of additional transient response within my working dynamic range. Even that does not concern me a whole lot because I use plenty of dynamic range compression, limiting, downward expansion and gating, effecting the working dynamic range of every track, virtually, anyhow. It still retains plenty of transient dynamics by running incredibly slow attack times and/or " no lookahead ".

Then everything will end up sounding really lovely and natural when it's actually anything but. And that's something we all had to practice at. In many ways, it was a great thing that we did not have the huge overwhelming selection of hardware and software as we do today. So with a much smaller cache of equipment, you'd definitely had the opportunity to get extremely deep and intimate, with each and every piece on each and every sound source, you could find, to try it out on. And when you found something that made some instrument or other sound source track, sound absolutely incredible, it still might not mix well with everything else. So what wasn't just the pursuit of getting individual tracks and sound sources to sound incredible, individually but actually, as a whole. And where some things that might not sound 100% pretty when soloed, just work right within the mix?

Today the selections are so awesome and extensive, that it could take you an entire year to make a decision on what each and every track might need? And while that's great with big budgets and lengthy production schedules... and it's not exactly applicable to your local band in need of a demo or vanity album/CD/intranet, release. Which is why I also push up a mix with little or nothing on it to begin with to establish the overall balance, flavor and direction as we start to polish up a mix.

Then once you get a good compatible mix that plays well on most everything, you'll be able to get ready for the final stage. The final stage of course being the Mastering of those final mixes. While a lot has been talked about and discussed about " self mastering ", I think it's absolutely sensible, applicable to at least give it a shot. This isn't a further attempt to at least get a feel for how this finished product, might compare to other similar professional released product? It might even make you realize that an additional multitrack, more final remixing might be necessary in an above your already finished mix? And then when you think you've got your mix down just right? You're ready to ship it off to the real deal, Mastering Engineer, who will take your product to the next level.

Exciting things, experience, knowledge and revelations are knocking at your front door right now.
Mx. Remy Ann David

And while that's great with big budgets and lengthy production schedules... and it's not exactly applicable to your local band in need of a demo or vanity album/CD/intranet, release. Which is why I also push up a mix with little or nothing on it to begin with to establish the overall balance, flavor and direction as we start to polish up a mix.

at what point do you include the bus comp eq and limiter? right away? after tracking? mixdown? not being a jerk, jw, ?

That's an excellent question. I can tell you that with my work, that varies. For certain jobs, live FM broadcasts for example, I've had requests for a stereo bus limiter on my master stereo output. I'm happy to accommodate that. And equally, I have to be able to monitor that. As with any limiting or compression, broadband in particular, it can screw with the overall tonality of the sound. This is where I'll bring in my/patch in my Orban, graphic/parametric EQ's, APHEX Aural Exciter, to pull out what I need to hear.

Conversely, on non-real-time, remix jobs, I developed the mix that I want without any of that. It's only after I'm done with the mix there I go on to do whatever it is I do, in software or hybrid style to complete the mix. Which a lot of folks would consider or classify as Mastering but it's not. It's not the intention. It's the attention I'm looking for. Upon my completion, I feel that it is appropriate then to go on to an actual Mastering Engineer, not myself. And my waveforms do not look like bricks as is common for most others to end up with. None of the attack times or release times I use in whatever processing be at broadband or spectral based, has anything on it other than ultra-ultra-slow attack and release times and no " Lookahead ".

The Mastering Engineer, may in fact yield a brick looking like waveform? But it will be the right sounding brick looking like waveform that one really cannot obtain from anyone's software nonsense or plug-ins by themselves. I know there is some excellent popular ones that do a wonderful job in the right hands of course. So it's not completely inappropriate to accomplish the mastering all ITB and even with a single software plug-in. Many of us prefer analog summing to the digital summing which makes everything sound like it was done on ProTools. But sometimes, that is the required or desired character of sound one wants to go after and there is absolutely nothing wrong with that. It's just another one of the many colors in our palate to use. I mean I'm old school but I have no problems doing everything ITB and with all digital consoles. Why shouldn't I.? I'm an engineer. An actual audio engineer that actually engineers audio and knows how to.

So when people tell me they want that nice huge fat analog sound, I can give it to them. When they want the best in digital, I can give it to them. If they want the best from both, I can give it to them. It doesn't matter. My final outcome is still quite consistent regardless of how or what I work with. And engineering styles, equipment and technique, I have to change accordingly to retain " my sound ".

For instance in a recording that I made in New Zealand, it was my intention to use my API 3124 mixers. It was also my intention to use my Beyer M-160, ribbon microphone, on this dramatic soprano, I had recorded in the past that way. But compromises had to be made because we were sharing the cathedral with the New Zealand government sanctioned New Zealand Radio Engineers, who were also making an operatic album in the afternoons and evenings, after us. So the setting up and tearing down of my rig and their rig was quite impractical. They wanted to use their microphones. My producer wanted me to use my microphones in a completely different set up than the New Zealand guys. And the compromise was the mixer I had to use. It was an AMEK, BC-1 very small portable mixer. And it was considerably darker sounding/smoother sounding than my rather aggressive API's. This forced me to change out that M-160, for an AKG 414 B-ULS, to achieve a similar tonality on her. And that's way different. It still didn't keep me from receiving a Grammy nomination for that best engineered female operatic, award nomination. And it was a huge compromise! Big deal! It all works. The meters move. That's all that matters. I'm certainly not a rigid recording engineer because of my broadcast background. It doesn't have to be just so. It only has to be professional and it is. I can roll with it. I can roll with anything. And that to me is the mark of a good and competent professional engineer. Not what plug-ins or equipment you use per se. It's whatever gets the job done the way the client wants it or the way I want it. Which is frequently the case because that's why people come to me. I'm usually just given free reign to do what I do.

This is all for one of the early digital esoteric labels. It was only done with the finest state-of-the-art equipment like my Panasonic SV-3500, SANKEN CU-41's, AKG 414, 87's, 86's, AKG C-451, SM 81's and all my M-160's/130. And back then, the Sony digital Mastering System offered up nothing more than volume changes and the timing of spaces. There was no actual Mastering and the recording was released as recorded. I thought surely they would have used some kind of limiter? But nothing of the kind nor any equalization changes were done. Syncing of the ripped CD and my backup safety DAT's, phase inverted and collapsed to Mono, leaves nothing behind! Holy crap! So not only did I record it, I obviously mastered it? Which was nothing more than the original stereo mix. And I was doing it on his bizarre soffit mounted huge control room Panasonic speakers I had never seen or heard of before. It had a rather fascinating looking solid woofer that looked like it was made from a beehive out of metal. So I was actually recording and mixing on monitors I was completely unfamiliar with LOL. No other references available other than headphones. This is all back around 1992 when there were no boutique high-quality converters of any sort. All first-generation 16-bit converters and only recording at 44.1 kHz.

So everybody goes about this on their own ways as to whatever works. I'm certainly not telling the client I could make a better recording if I had better converters. No way. And at the time there were no other choices. So everybody was happy with what they got from those first-generation converters of the average type considered professional. I mean even back then CDs sounded like we were delivering master tapes at 30 IPS in comparison to the cruddy sound of vinyl. Early digital even transcended producing at 15 IPS on the MM 1200-24's anymore and most of us went 30 IPS. And that was back in 83. Sure I kept recording at 15 IPS for those that could only afford to record at 15 IPS. I actually did some albums that were only 12 track recordings. Because we can only use a single roll of tape that the band could afford. So it was 12 tracks times 2, on a single reel tape.

I think Mastering, as we know it today, has become something of a cult favorite for fun, for a lot of people? Everybody wants to take their favorite rock 'n roll sounds from their favorite rock 'n roll groups and make them better. It's like playing a video game with music. And then they call themselves Mastering Engineers when they really know nothing about it other than their cracked plug-ins on Cube Base. Along with their super high quality Event 20/20 monitors which we know to be extremely accurate for any Chinese rock 'n roll. Especially the electronic stuff. Nothing sounds better... ain't that the truth.

Glad I don't know those monitors
Mx. Remy Ann David