This shoot was a small part of the documentary “Zero Days Until Launch” produced by MTSU in partnership with Nashville Public T.V. The documentary is in post-production and should air sometime in the early part of this year. This was a multi-cam truck shoot with a crew of 20. As A1, my job was to record audio of the launch of the space shuttle Atlantis.

Check out the mixonline article about the recording..

America – Anniston, AL

Freelance – 2008

America - Anniston AL

12 Photos

 

I recorded this show in exactly the same as I did on Fiddler on the Roof. A full write up would be rather redundant, so please, enjoy the sights and sounds. Special thanks to Jeff Worrell, from America’s production team, for helping make this recording possible.

View From Monitor World

RagTag Quartet

3 Photos

 

The RagTag quartet is a traditional Barber Shoppe quartet from Nashville Tn. We created these3 recordings at MTSU in the class Production Seminar I, which is first year MFA. We chose the RagTag Quartet because it was something different. Most everyone records rock or pop music around here.

Two approaches, or production methodologies, were considered leading up to the sessions, an ensemble approach and an individual approach. The ensemble approach would utilize stereo microphone techniques to capture the whole quartet to a pair of tracks. The individual approach uses a separate mic for each person in the ensemble recorded to separate tracks. Both have their various strengths and weaknesses. Current production trends, especially in the studio environment, would often dictate that we take the individual approach which would allow us to exercise the control, and propensity for non-committal processing, allowed by computer recording technology.

In the end, we decided to opt for the ensemble, or stereo, approach which poses the new question of which stereo technique to use. The first stereo techniques that come to mind are the spaced pair and the coincident pair. The spaced pair would offer a wide stereo image and poor mono compatibility. The coincident pair would would have a more narrow stereo image, although still very stereoy ( stereoy – that the best I could do), and be more mono compatible. We could use the Mid/Side technique that usually yields an excellent stereo image, is mono compatible, and would allow us to make some changes to the balance of he ensemble. Or finally, the Blumlein pair which would allow a greater sense of space due to the figure eight patterns of the mic.

What to do….

Try em all!!!!!

In the bottom left we have our M/S pair and almost directly above that is the Blumlein pair. To the outsides is our spaced pair, which is a little more spaced than I would have normally liked, however space is a concern. Finally, towards the top right, on the stereo bar, is our coincident pair. First, let me clarify that only two mics in this situation would ever be used from this amalgamation of stereo setups. Trying to use more than one pair at a time on the same source could be vary bad. Also of note, is that the M/S decoding was handled inside and outside of pro tools by routing the side mic to two tracks in pro tools, and reversing the phase on one of them. We then sent the tracks to three channels on the SSL Duality and took are of the panning there. The MID is panned dead center, SIDE 1 (no phase change) panned hard left, SIDE 2 ( 180˚) panned hard left. This allows us to monitor the M/S signal decoded, and in confidence.

Trying to figure out which technique was our favorite for this situation is where things got interesting. In the end, we agreed that the best stereo technique on tracking day depended almost exclusively on the song. Obviously there would be major differences in the sound of each stereo pair, but should not the ensemble sound relatively the same from each song to the next on the same pair of microphones?I thought it should, and I was just plain wrong.

Have a listen. Each of these example are unprocessed, and represent the input directly. I find one to be more appropriate than the others. What do you think?

So what is the variable? The only explanation I can conceive is that the song is the variable. Every song is different and, as it should be, a musician approaches every song differently, giving it the shape and balance that it calls for. That being said, we should be able to construct a set of guidelines that would indicated which pair would give the best results. However, for a “type” of song there are an infinite amount of artistic possibilities that would prevent the creation of any hard and fast rules. The final judge should be the ear and which one feels the best. The choice can have a profound impact on the experience of the final listener.

The mixing phase was easy because largely the work had been done on the front end. The group balanced themselves well and their where very few edits to make. All we really had to do was add a little reverb since the space we recorded in was very very dry. Other than that we ran the tunes through the bus compressor on the SSL AWS900 and we where done. We mixed all six songs in about 4 hours and where happy with four of them. The Total time spent on the project was around 8hrs.

]]> http://adamdellis.com/?feed=rss2&p=444 3 http://adamdellis.com/?p=397 http://adamdellis.com/?p=397#comments Sat, 22 Aug 2009 19:51:20 +0000 admin http://adamdellis.com/?p=397 Recorded at MTSU Studios A and B – April 2009

Austin Ellis

12 Photos

 

Session Report

Man I dig this stuff! Austin and his band have a very smooth, organic hip-hop kind of sound, which can be fun from a production point of view. Not only can you work from the perspective of purity, or recording natural sounds and performances, but this is a wonderful opportunity to combine that perspective, with electronic production techniques. Our goal is to capture the natural feel of the band playing, and maintain a sound that resonates with hip-hop overtones. The instrumentation for these recordings consisted of drum kit, electric bass, electric guitar, acoustic guitar and vocals.

In the planning stages we decided that we wanted to capture as much of a live performance of each of the three tunes as we could. Once we has a solid performance, we could then go back and massage the songs into their final versions via editing and overdubbing. However, everyone involved had day jobs and the studio time was booked at times the space was available. I believe there was a 12am – 8am session in there. In the end we basically had to make progress on each song at the mercy of what, and who, was available at the moment.

In the first tracking session only Austin, our front man and principal songwriter, was around for the first four hours. Using a click, we tracked his acoustic guitar and fortunately, Austin was very well prepared. He managed to lay down the tracks near perfectly. Although the parts were very simple musicly they did lay down a triffic framework on which to build the rest of the parts. In only took a couple of hours to get Austin’s guitar parts done, and with some extra time on or hands, we decided to go ahead and cut vocals with the idea that we could use them, or replace them later. We had to make the most of our time and in a couple of hours we excelent guide vocals, if not usable tracks. Keep in mind that we have stuck to a strickt musical structure and so far everything has been recorded to the click as tightly as possible. If the guide tracks are air tight we will have fewer problems getting the rest of the band in there.

The rest of the band members made it to the session and the plan is to track everyone together. Studio A at MTSU has a large tracking space, a small ISO booth, and a large selection of absorptive and diffusive gobos which makes isolating the various instruments, while maintaining sight lines for musicians, pretty easy. To record the drums we used a simple, augmented stereo, four microphone approach. Two Neumann TLM107 microphones were placed about four feet out from the kit, and about three feet off the floor in a 60˚ triangle. Note: We did not measure the triangle because close will work just fine in this case. Next we mic’d the snare with an SM57 placing the edge of the capsule in line with the snare shell vertically, and 90˚ to the head horizontally. A Sennheiser e602 was used in the kick drum, pointed directly at the beater and placed as close to the batter head as possible.

In order to shrink the acoustic space, a really large room, the whole drum setup was placed a few feet in front of a concrete wall and two large diffusive gobos placed to the coincident side of the TLM107’s. Our space still sounds big because we are still getting a lot of early reflections in the sound, but the drums wont bleed all over everything else, and with some luck everything else wont bleed all over the drums.

The drums themselves sound fantastic, which is 99.99% necessary for this approach to work, but back in the control room things are sounding a little weird. There is a phase issue going on between the kick and snare close mics, and the room mics due to their relative distances from the drums. At editing time, this will be the first correction that we will make by aligning a single snare and kick hit to the same instance in the room mic tracks. This will correct the timing difference, due to distance, and thus the phase problems resulting in a crystal clear ambiance.

All of the mic lines, and the bass DI, where recorded via Millennia mic pres that where connected to the Studer D950 consoles A/D converters, then directly into Pro Tools. Retern monitoring paths where also handled on the D950. We took many, many, full band takes, and even though they where all cut to the same guide tracks, every take was unique, sometimes to the extreme.

The Mix

The largest and by far the moist complicated part of the mix phase is editing. We have to evaluate all the takes and build the final version of the song. There is a lot of opportunity here for an engineer to be creative but we have to remain catious in order to maintain the feel of a liner and natural performance. However, if a moment or a phrase seems a little mechanical it might lend to hint at the hip hop influence.

Mixes of the songs where done largely “in-the-box,” on my macbook pro with a Mbox. Because we payed extreme attention to the sounds we where getting at tracking time, and how they where blending together, very little processing was needed. The process was mostly about balance and panning, and making sure the song was musically interesting.

At the time of the tracking sessions, Austin and his band had only played together for a very short time, and where only familier with the material. This type of situation could be disastrous, but one again great work ethincs and attitudes saved the day.

Here is a picture borrowed from electronic musician.

The stereo image is produced by manipulating the phase differences and similarities between the two sources.  But first, lets dig into our knowledge of basic acoustics and remember that two otherwise identical sound waves that are 180˚ out of phase will cancel each other completely. Acoustic annihilation! If the sound waves are only partially out of phase there would still be some cancellation, rather a net loss of acoustic energy. If the two waves are completely identical and in phase the would sum, for a net acoustic gain.

As we can see from the picture above, sounds originating from a frontal source would arive at the two microphones at different times, thus puting the electrical signals generated by the microphones out of phase.  We also have to consider the indirect sounds, reflections off of various surfaces in the room, that are arriving even later than the direct sound that give us our sense of space.

To take advantage of the various phase interactions, and produce a stereo image, we need to sum the Mid and Side, and send this new signal to the left channel. For the right channel we need to sum the Mid and Side again, but this time we need to reverse the phase, or flip the signal 180˚. The effect is like taking a positive number and adding a negative number to it…

Left = M + S

Right = M + (-S)

The center of the image is of course regular ole stereo phantom center and, magically, it consists of the sound source that was directly in front of our M/S pair.

Since we are building a stereo image out of the various phase information our microphones have captured, there is no reason why we could not deconstruct a stereo image via the same kind of phase information, manipulate or change it, and put humpty back together again. Mastering engineers do this frequently in order to apply processing that they do not necessarily want to affect the entire mix. For example, a mastering engineer is given a a mix of a pop song, and for whatever reason the vocal is way to loud, or is very sibilant. Since it is a pop song, we can assume that the vocal is panned dead center. The solution may be to use M/S processing to separate the stereo recording into Mid and side channels, and then simply lower the level of the mid channel, which is the center of the stereo image. When its put back together into a new stereo recording the vocal will be quieter, but so will everything else that is panned dead center. If the snare was in the perfect place in terms of balance we may have created more problems in our attempt to fix the vocal. Still, if lowering the level does not do it, try a multi-band compressor, an EQ, or anything else your heart desires. One more thing to note is that when you lower the level of the mid channel you will also increase the perceived width of the stereo image, and vice versa.

There are alot of ways you can go about acheving the M/S process including a variety of off the shelf plugins that just do it for you. If it makes your life better, please,  go that route. However, you can achieve the same, if not better results, with the fetures provided in the most modest version of Pro Tools. Although this article uses Pro Tools as an example, I feel certain that you can pull the same thing off with most any multi track D.A.W. platform that allows internal busing.

Once you have your session created in Pro Tools you will need to create one stereo audio track, and five stereo aux inputs., then label and color code the tracks as follows.

1. (ST audio track – Blue) “Source”
2. (AUX input – Green) “Mid”
3. (AUX input – Green) “Side”
4. (Aux input – Yellow)   “Left”
5. (AUX input – Yellow) “Right”
6. (AUX input – Red) “Decoded

You can use any colors you would like, just make sure to group the tracks as we did above. Next insert one of the time adjuster plugins – under insert–>Multichannel Plugin —> Delay —> Time Adjuster (short) – into each of the five AUX tracks..  You can use any stereo plugin that will allow you to reverse the phase (ø) of either the left or right channel independently.

From here we need to begin to route our tracks to their appropriate destinations, and this is where it can get a little tricky at first.  However, we have already arranged our tracks so the signal flows basically from left to right. Start by assigning the “source” track to “bus 1 – 2.” Next, set the input of both the “Mid” and the “Side” tracks to the same bus. Now, your source track should be feeding both the “Mid” and “Side” tracks.  Now, assign the output of the “Mid” track to “bus – 3.” Please not that you are assigning a stereo bus to a mono output, which results in the summing of the stereo input, or to put it into language we understand, this is Left + Right = M.

Next assign the output of the “Side” track to”Bus – 4,” again summing the channels of the stereo source track. However, this go around we will use the time adjuster plugin to reverse the phase (ø) of the right channel. This equates to Left – Right = S. If you’re still with me, this should look an awfull lot like the expressions M+S = L, and M – S = R. Essentially the same thing is happening.

For our “Left” and ” Right” tracks we will set the input to “bus 3 – 4,” the outputs to “Bus 5 – 6,”and pan both sliders for each track left and right respectfully. Next, via the Time adjuster plugin, reverse the phase of the right channel of the “Right” track. Essentially, we are taking the M and S and re-encoding them to a new stereo pair.

Finally, set the input of the “decoded” track to “bus 5 – 6″, and the output to “interface 1 -2.” This is not really doing anything beyond providing a single fader for the final stereo output. You will find that piling all these tracks on top of one another will require to you bring down some of the starting levels. I usually begin with the source tracks fader at -3, and the Mid, Side, Left, and Right at -6. I generally leave the “decoded” track at unity.

Notice that even though we inserted the trim plug in all but the source track we only used two of them. Plugins, regardless of how hard they are working, will add latency to the signal, which will cause phase shift. The effects are usually very painfully obvious. Give it a try by removing the plugin from either the “mid” or the “side.” Also, if you command – click the numbers below the faders you can cycle through information such as peek level, fader level, and latency which is represented in terms of samples. Our first track should read zero, and the rest should read about 4, meaning that these tracks are about 4 samples behind tracks with zero latency. Regardless of what it is, if tracks 2 – 6 have a matching number, you’re good.

Now you can import the stereo track of you choice, and if you have never heard it before, prepare to be amazed. Have fun. But while going crazy and pushing the limits can yield some very productive results, attempt to make subtle adjustments once you get it out of your system. Try to bring impact to the music, or change the blend of the instruments, while leaving the sonic quality intact. If you are using a commercially released album you may be hard pressed to improve it. Still, breaking down a stereo mix into its true anatomy can be very revealing. I find myself in mix and tracking sessions often using the perspective I gained from just playing around with M/S processing , and undoubtedly my mixes improved.

And now that you have been through all that, here is my simple thanks to you.

Free Download: M/S Processing Template for ProTools 7.4

Now go play!

Adam

Fiddler

13 Photos

 

The audio recording of Fiddler was part of my final project at Jacksonville State University. It was actually two classes; Special Problems in Drama + Sr. Practicum = six credit hours of independent study. The result of the classes is a minimum 40 page paper detailing the plan, the execution, and the experience.. The first part of the project was to handle all of the live audio needs of the production, and second to make a multi-track recording of it. One of the first challenges is to find ways to accomplish both goals with one engineer – myself.

Jason Bozman - Gadsden AL

Even though the recording of the show is forever and the experience repeatable, the live audio production is paramount to the recording. One reason is that we only get one shot at any given one of the fifteen or so performances. For the recording, only one of those performances has to be a keeper. But perhaps the most important reason to give priority to the live audio production is the audience, who paid good money to be there.

The sound reinforcement needs of the show are rather substantial and requires quite a bit of attention during each run. Our input list consisted of about 24 channels of wireless “face mics” using Countryman E6 and B6 microphone elements and a 14 member pit orchestra. Also, we used a couple of microphones on each side of the stage in the wings. Performers who were not onstage, or about to make an entrance, sang around these mics to provide reinforcement to the chorus component of the ensemble. As a side note, we used a closed circuit video system to provide a view of the conductor backstage. In the pit were two percussionist, double bass, 2 keyboards, 3 trumpets, 2 trombones, 3 woodwind players with various instruments, and a sad violin.

The theatre seats nearly 400 and is very intimate making the sound reinforcement needs unique. The composition of the orchestra is designed to meet the minimum requirements of the score, still the orchestra will most always be too loud for such a small theatre. The face mics are used to add just a little kick to the performers voices to balance the pit. Usually very little coverage of the orchestra is necessary using only a DI for the keyboards for the onstage and backstage monitors, and a bass DI or mic, to help support the overall ensemble sound. In this instance however I added a series of spot mics, one for each section and an overall stereo pair. The intention was to use these mics for the recording only, but I did find them useful at times in the house.

One of the big goals, that is always present when doing sound reinforcement for a musical, is transparency of the sound systems influence. We don’t want the audience to know we are doing anything at all, and go to great pains to conceal our physical presence, and the presence of our work. Often this requires careful time alignment of the various components of the system, to ensure the voices appear to be originating from the mouth of the performer, even though it would be physically impossible, for these singers, to balance the orchestra without a little help. However, that is a long discussion for another day. Still, the topic is covered in detail in the paper I wrote documenting the project, as is the entire sound system alignment procedure as well.

MY16AT ADAT I/O (bottom)

Now that we have touched on the requirements of the sound reinforcement part of the project we can consider what we need to do to record each performance. Since the house mix will require nearly all of my in run attention, the recording system needs to be practically self sufficient. Because this is a university show, my equipment choices where limited to what was available at the school, and whatever I could acquire on my own. That is not to say the inventory at J.S.U. is lacking in any way. In fact, it is quite nice and includes a Yamaha LS9-32, 20 channels of Sennheiser wireless, and an adequate F.O.H. speaker system.

The LS9-32 has the ability to directly output each input channel to any output on the console, from anywhere in the input channel’s signal path. Because the LS9-32 is limited to 16 outputs, it was necessary to add two MY16AT expansion cards giving us the necessary number of outputs to cover our speaker system, and the “1 to 1″ recording of every input channel. The MY16AT transmits its I/O in the ADAT format which is convenient because many computer audio interface’s support it.

At the time the best option we had for adding 32 channels of simultaneous I/O to a computer was the M-Audio Lightbridge.

M-Audio Lightbridge

The computer for the gig was a monster Intel quad core PC running Sonar which ran flawlessly for fifteen performances. The computer recorded continuously for nearly 2.5 hours each night and never thought twice about it. One reason the DAW was so stable is that the PC was not worrying with AD conversions just shuffling the information from the console to the hard drive in a manor humans can make sence of. The primary recording drive, by the way, was USB 2.0. – seriously – and it worked just fine.

Direct outputs where taken from each input channels post AD conversion and before any of the channels processing resulting in 32 tracks of raw input. Each night, I have to hit record and make sure that no inputs are clipping. Since all of our meters are post AD, clipping will occur at the same time on the channels input as it does on the recorders input. If for some reason the two clip asynchronously Yamaha has provided a “direct out level” to compensate. This makes the rig very low maintenance and easy to monitor.

F.O.H. Position

In the Mix

The largest body of work in the post production process for this particular situation was selecting which performances to use, and editing them. Fifteen performances times 2.5hrs each equals 37.5 hours of audio that has to be sorted and evaluated. What I was looking for was basically a performance of a particular song that contained no “deal breakers,” which is any snafu that cannot be easily edited or covered. For example, a snafu could be a performance issue like intonation, major timing errors in which the ensemble falls apart for a period of time, or perhaps someone missed and entrance. Maybe the snafu was a technical issue like RF interference in the wireless system, or a faulty microphone element, which happens very frequently despite our best efforts. So the goal is to find the best performance of a given song and the best recording of it. Given all the human elements involved with a production of this scale, out of fifteen performances I ended up with one or two choices per song

The DAW was recording all inputs continuously, regardless of whether or not the performer was onstage, in the wings, or in the bathroom. ( They were well warned and besides – that’s show biz). I find that 90% of mixing music is preparing to do the mix. Organizing the tracks, creating and routing buses, setting up effects, and cleaning out all the garbage is a lot of busywork, but if you slack off here, the results will not be anything like they could be. The creative stuff simply cant happen, for me anyway, until the tracks are all cleaned up and a work flow is established.

I began mixing by working on getting the orchestra straightened out first. Once I had a basic mix of the pit I then started working on getting the vocals together. I found blending as many as 20 individual voices a very time consuming thing to do. A good ensemble blends itself naturally with each part adjusting itself in relation to all the other parts continuously. What this means for me is a lot of automation. I don’t have to recreate the natural balance and blend entirely by artificial means. The adjustments made by the performers are still there. What is missing is the point of reference to which the performer was adjusting. In short, we don’t know exactly what the performer was hearing, which is exactly the information used by the performer to blend naturally. To make matters a little more difficult, because the performers do not stand still on stage, that point of reference is constantly changing.

The first step is to organize the vocals by leads and chorus, part, and into male and female groups. Then I started to build the vocal ensemble from the bottom up against the orchestra. Once a general balance is achieved I used automation to adjust for any balance issues that occur over time, probably caused by the performer trying to balance within the ensemble he or she was hearing. This is not necessarily difficult work, but it does take patience.

It is important to note that the intention is to not to create the best possible performance in post, but to make the best performance sound a good as it possibly can. A little dirt never hurt anyone and after all, its live.

To Life!!!

]]> http://adamdellis.com/?feed=rss2&p=303 0 http://adamdellis.com/?p=290 http://adamdellis.com/?p=290#comments Fri, 14 Aug 2009 01:09:14 +0000 admin http://adamdellis.com/?p=290 Recorded at Trevecca Studios – January 2009.……………….

The Slips Session Photos

33 Photos

 

Six songs were recorded over two marathon days in January of 2008 at Trevecca Studios on the campus of Trevecca Nazarene University. They have a fantastic recording studio at the school even though some may see it as modest. You will not find an Solid State Logic or Trident console, 24 track two inch machines, or even a ProTools HD rig. However, you will find a very well planned and cost effective compliment of equipment. Simply put, the gear selection is down right smart.

One of the smartest things about Trevecca studios, and what manager Blake has done with the place, is the selection of seriously high quality preamps and converters. Available outboard preamps where API 500 series, Vintech, Drawmer, and AEA amongst others. The console is Mackie 32×8 which is used strictly for monitoring. His studio there is based on a DIGI002 system, however, Blake has armed the control room with an Appogee Rosetta 800 converters. For these sessions we brought in a DIGI003 because it has wordclock I/O, allowing us to take advantage of the Rosetta’s more stable wordclock. Of course, on a 002 you could clock to the ADAT from the Rosetta. None the less, why not use the latest and greatest if you have it.

Chuck in the tracking room.

The crowning jewel of the studio is the space itself. The main tracking room is small but sports floating hardwood floors, adjustable acoustic paneling (diffusive / absorptive), and there are wood surfaces everywhere. Over all the sound of the room is fantastic even though most of surfaces are highly reflective. This hyper reflectiveness can be toned down a bit with the adjustable paneling but the room still sounds larger than it actually is. I found this contributed greatly towards getting a big drum sound.

In fact – why don’t we go ahead and talk drums.

First and foremost, and the most important part of getting a great drum sound, is great sounding drums. Rocket Surgery – I know. Still, this aspect if often overlooked by many. Here is drummer Chuck and myself micing up the kit which sounded fantastic.

All of the microphones we used on the drum kit are fairly standard issue. On the kick we used a D112 and on the snare top and bottom we used SM57s. On the toms we used Sennheiser e609’s and the overheads where Neumann KM184’s. Like I said, pretty standard drum tracking fair, however, we did employ one additional technique. We placed a pair of AEA R92 ribbon microphones, very haphazardly and almost at random, in the room kind of pointed at the drums. We just wanted one room mic on each side and a little in front of the kit. We then ran the R92s to a pair of AEA ribbon pres and then squashed the beejezzus out of the the signals with a tube compressor, the model of which I cannot remember.

When we mix the overall drum sound with the super compressed room sounds the result is kind of like parallel compression except that the two sources signals do not match exactly. In the end you are mixing the close mic’d drums with a hyperized room sound giving you a degree of control over the amount of room sound that is in the mix and therefore adding a great degree of depth, in my opinion. In the final mixes we had no need to use any reverbs or other time based processors on the drums and the less processing the better.

One more thing that contributed to the fantastic drum sounds that we got….

Drummer Chuck played his ass off. As a side note, The Slips usual drummer was MIA at the time of the session and Chuck stepped up and nailed the parts even though he had never played them before. As I understand, he learned most of the tunes in the car during their trip to Nashville. I look forward to the opportunity to buy this man a beer.

What we usually think of next is the Bass, but unfortunately there is nothing to special about how we recorded the bass guitar. All we did was record it direct though a Vintech preamp, which was rather nice. The electric guitar sound, however, where are personal best. Here is half the cause….

This is a Dr. Z Carmen Ghia. As you can see, it has a plug, volume and tone knobs, a little blue light, and a switch. It is 15 watts of raw power and a thing of immense beauty. We tried several really nice microphones on the amp, including some seriously expensive ones, but finally settled on this Sennheiser e609. We dropped the mic on the speaker old school style, and ran it into a Drawmer 1960 tube preamp / compressor. At first it sounded pretty damn wimpy but we found that the more we cranked the amp, and the more we let the Drawmer do its thing, the sound got better and better.

Recording with compression on the way in is usually a very big no no because capturing raw unprocessed sound for later manipulation leaves all your options open. However, I don’t see the logic behind putting down a bad sound for any reason, especially a fear of commitment. So in dialing in the guitar sound I abandoned all needles and meters except the guitar track’s input meter. At this point there are two goals. One to get a great sound and the second to not clip the input of the guitar track.

The louder we got the Dr. Z and the harder we drove the preamp, the better the sound got. Just for fun we took an SPL reading in the room once we got it dialed in and measured 125dB SPL. Not bad for 15 watts!

When we did decided to see what the meters (input, output, and gain reduction) on the preamp/compressor were doing we found that all where just about buried anyway. You zoomed out on the waveform for the guitar it looked like a solid bold line, and zooming in a bit revealed a visual representation of the compressors time attributes. Normally this might not be a good thing but the ears are king, and this sound would clearly work for us.. The lesson? – Sometimes abandoning convention and theory, especially when it is not getting you were you need to go, works to your favor.

Dave Harper

To record mandolin and resonator for a couple of the tracks we use the “Mid-side” technique. For the side we used and R92 and the mid we used a KM184. The resulting sound was very articulate yet smooth and warm. The R92’s frequency response is not the same on both sides of the ribbon and each side has its own unique color of sound. The difference is subtle but is very noticeable especially over time and I would say that one side of the mic sounds a bit brighter than the other. This variation of color makes for a compelling sound on the M/S decode – blend to taste.

Vocals. We used a variety of microphones on vocals and the decisions where largely dictated by the track at hand.

Rob Davis

Rob Davis ladies and gentlemen! He’s cutting vocals through one of those AEA R92s’ – Dig it on the vocals for the acoustic tunes we recorded.

We also used a U87 and a Rode Classic through a Vintech pre.

Rob and the U87

The Mix

One of the greatest things about modern recording technology is the ability to do alot of work outside the studio. All of the Slips stuff we recorded were edited and ruffed mixed largely at home. For final mixes we chose to employ a hybrid digital/analog setup that yeilded the full power of both worlds. At least where mixing is concerned anyway.

MTSU Studio C

We mixed the songs at MTSU’s studio C for the most part, with the exception of Cinderella Blues which was mixed at MTSU’s studio B. Studio C has a ProTools HD rig, SSL AWS900 console, and a nice compliment of outbard compressors and reverbs. In the box we grouped the instraments in a familiar fasion and bussed the group outputs to the SSL. Tracks that we wanted to insert a piece of outboard gear into were also sent to the SSL as individual channel. When the SSL is in analog focus we have mixture of stereo busses, e.g Drum, Gtrs, and VOX, and inividual channels like “Bass guitar”, and Lead VOX on the faders. In DAW focus the faders represent the individual tracks in ProTools. Once we have everything balanced we mixed down the song to a stereo track back into ProTools. The greatest benifits where the analog summing, the use of the outbard EL distressor on the lead vocal, and the SSL Bus compressor, which really kicked things up a knotch.

And in the end…

So how did we manage to do basic tracks and overdubs for six songs in less than 48hrs? First, everyone involved had an excelent work ethinc and attitude. Reguardless of how the recordings turned out, which I am very pleased with, we had a great time. Second was the fact that the band was well seasoned enough, and as prepared as they could be, to nail everything in just a couple of takes. Finally, the studio itself functioned near flawlessly.