STUDIO SOS

Nick Tucker

Published in SOS October 2002
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Technique : DIY

The boxiness of Nick Tucker's vocal recordings was tackled by adjusting the mic positioning and by holding up a duvet behind him to damp down excess room reflections — anything to keep Editor In Chief Paul White out of mischief...
The SOS team rushes to the rescue of a reader in Somerset suffering from boxy vocals, a weedy mix, and a dodgy tweeter...


Paul White

After reading the first of our Studio SOS features, Nick Tucker sent us an email entitled 'I Want One Too!' and suggested that we might be interested in sorting out his vocal recording and mix problems. So Technical Editor Hugh Robjohns and I arranged to visit, taking along a few bits and pieces of studio gear which we thought might help out.

Nick is a Barrister specialising in criminal defence, but has reduced his work load in order to allow himself more time in his studio, which, unusually, is in his girlfriend's house — at least she knows where he is, I guess! The house is in fact a converted school located in a rather picturesque part of the Somerset countryside and Nick's studio is in what used to be a cloakroom. This presents a challenge right away, because, although the room is some 15 feet long, it is only around five feet wide, so Nick works across the room and has all the equipment and monitors mounted low down so that he can work while sitting on the floor.

When Nick first called us, he said that his vocal recordings had a somewhat boxy quality, despite being recorded using a Neumann TLM103 microphone, and he was also a little unhappy with his mixes insomuch as he felt the sounds didn't gel properly and were also lacking the clarity of commercial mixes. To solve these problems, we needed to look both at the recording chain and the physical way in which the vocals were being recorded.

Mic Positioning

At the heart of the studio was a Tascam 388, which is a kind of giant Portastudio with an open-reel eight-track analogue tape recorder built into the mixer. It uses Dbx Type I noise reduction, and Nick employed a separate Behringer mixer to combine the output from the

A fair amount of high-frequency information was being lost by Nick's old Tascam 388 — a brief inspection revealed that the heads were quite worn — but there were also problems with the Dbx noise reduction.
388 with that of a drum machine and a selection of effects units, including a rather nice Lexicon PCM90 reverb processor, an Aphex Model 109 parametric equaliser and a couple of TL Audio equalisers.

Nick describes his music as pop with a folk influence, and he both sings and plays guitar. In the case of the acoustic guitar, he was tending to use a mixture of DI and microphone, whereas electric guitars were often DI'd via a Line 6 Pod and bass parts via a Line 6 Bass Pod, which Nick is particularly fond of.

For vocal recording, Nick was setting up his microphone at the opposite end of the room to the equipment, in a corner that he had treated with acoustic foam tiles. The TLM103 was kitted out with a pop shield, but no shockmount, and a Joe Meek VC3 preamp/compressor was in the signal path between the mic and the 388 recorder. To allow us to assess the extent of the problem, we had Nick redo the vocal to one of the tracks he'd been working on using his normal working method. Sure enough, it sounded a little boxy, but it didn't take too long to pin down the causes. Firstly, Nick was standing rather further from the mic than was ideal for such a confined recording environment (around two feet), so the room resonances were intruding onto the recorded sound. To compound this problem, two acoustic guitars hanging on the wall were vibrating in sympathy with his voice and adding to the unwanted resonances. This latter situation was quickly fixed by removing the guitars from the room. At the same time, we reduced the mic distance to around nine inches and moved the pop shield to within a couple of inches from the mic. Note that you shouldn't place a pop shield right up against the mic body — they work much better if there is a reasonable gap between the shield and capsule of around a couple of inches.

  All About Dbx Noise Reduction  
  It seems hard to believe now, but at one time the type of noise reduction process you used was as vital to the success of an analogue multitrack recording as the size of your hard drive is today! Professionals enjoyed the relative merits of noise reduction systems such as Dolby A (or the much improved Dolby SR later) or Telcom C4, while the home studio tended to use equipment fitted with Dolby B or Dolby C (and a few late models with Dolby S), or with one of the Dbx formats (Type I or Type II).

The basic idea of all these systems was to reduce the dynamic range of the input signal in some way prior to recording to tape. On replay the inverse process was used to restore the original dynamics (more or less) while also reducing the apparent tape noise into the bargain. The success of such systems was variable, though, and the best (Dolby A initially, and subsequently the remarkable Dolby SR) are technically complex and tackle the audio in several independent frequency bands to maximise control and minimise modulation side-effects.

The Dbx noise reduction system was relatively simple in comparison and, although not so widely used in professional circles, was adopted readily in the semi-pro markets. These Dbx systems, developed by David Blackner in the early 1970s, essentially employ a wide range 2:1 compressor on the record side, with a complementary 2:1 expander on replay. Since the 2:1 ratio is used over the entire dynamic range of the input material, there is no critical threshold requirement, and so no need to accurately match record or replay levels when tapes are moved between machines. This has always been one of the drawbacks of the Dolby systems, and this is the reason why Dolby tone is recorded at the start of multitrack tapes in professional studios.

In any broadband noise-reduction system such as Dbx, dynamic changes caused by a low-frequency source — a bass solo, say — will result in the level of tape noise being modulated, and with no high frequencies in the recorded signal to mask it, high frequency tape noise will become very obvious. To overcome this significant side effect, the Dbx system applied pre-emphasis (high-frequency boost) on the record side and a corresponding de-emphasis on replay. The idea was to reduce the audibility of high-frequency noise generally, and so the pumping artefacts of the companding process were less audible.

On a good day, with everything set up perfectly, the Type I system was capable of providing as much as 30dB of noise reduction — far more than Dolby A — although it was prone to audible side effects. The difference between the Type I and Type II formats comes down to the details of the level sensing circuits used to control the compression and expansion processes. Type I was optimised for high-quality, wide-bandwidth open-reel recorders, whereas Type II was designed for the rather less reliable results obtained from cassette recorders. Consequently, Type II noise reduction was not as powerful, and more prone to mistracking side-effects.

Whereas the Dolby systems only compressed low-level signals which were in danger of being lost in the tape noise, the Dbx system compressed everything regardless of level and, although far simpler and cheaper to make, this was also the system's primary weakness. If the recording medium was entirely linear at all levels, a 2:1 expander would always undo the dynamic changes applied by a 2:1 compressor, so the replayed signal would be identical to the recorded signal (ignoring the problems of compressor overshoot and the like). However, as we all know, recording tape is anything but linear, especially at higher levels where tape saturation occurs. The saturation effect means that a high-level input signal is replayed at a lower level and with a different frequency response (and more distortion). Since the expander can only work with the signals found on the tape, the original (but saturated) high-level signal ends up being decoded incorrectly. This is known as a 'tracking error' since the dynamic reconstruction of the expander fails to track that applied originally by the compressor. Tracking errors can also be caused by drop-outs, an azimuth error, or if the machine is not equalised correctly for the tape being used, and typically result in a dull and dynamically compressed (or 'choked') sound.

So, whereas the usual advice is to drive analogue tape fairly hard (even when using Dolby noise-reduction systems) to obtain the best noise performance and the often desirable characteristics of tape saturation, try this with a Dbx system and you will be badly disappointed! For this reason, many recorders that use Dbx have switches to disable the noise reduction on some or all channels, so that good old-fashioned tape saturation can still be used creatively if required. It is also a good idea to switch the Dbx off if recording transient-rich instruments such as percussion, since the companding may mistrack on fast transients.

When Dbx noise reduction is being used, it is generally wise to err on the side of caution when setting levels. Keeping peaks below the point of saturation gives the expander the best chance of reconstructing the original signal properly. This usually means 0VU is the absolute maximum level, rather than the average value to aim for, and recording at a slightly lower level than you would without Dbx isn't really a problem as the amount of noise reduction is so great anyway. It is also vital to keep the tape heads clean and demagnetised, and to have the machine serviced regularly and aligned correctly for the type of tape you are using, to ensure that what goes in comes out properly. Hugh Robjohns

 

Gain Management & Signal Processing

I also asked Nick to set the recording levels as he normally would, because I had my suspicions that the Dbx noise reduction had something to do with the problem. As I guessed, he was setting the levels a little on the hot side and, while most a

The SPL Channel One's Air EQ control was used to compensate for the high-frequency losses incurred because of the Tascam 388's worn heads.
nalogue tape likes to be 'pushed into the red' (to add a little warmth and compression), this has an adverse effect on the way Dbx noise reduction functions, as any nonlinearities in the recording caused by tape saturation are magnified by the rather extreme encoding and decoding system used by Dbx. For more information on this subject, have a look at the 'All About Dbx Noise Reduction' box.

Having corrected the mic distance and recording levels, we tried another take and found the boxiness much improved, though the vocals still lacked the airy presence that Nick was after. We also felt the 'room boom' could be further improved by placing some kind of acoustic absorber across the room, between Nick's vocal position and the part of the room housing the equipment. Once again we improvised using a duvet, and straight away the sound got noticeably cleaner, as almost all the remaining room resonances and 'colour' were isolated from the mic.

Having got this far, we decided to try the recording again using my SPL Channel One mic preamp, simply because it sounds very open and neutral, and also because it has a good EQ section and compressor. The Joe Meek VC3 is actually a very nice piece of kit, but it is built to a price and consequently loses out in the metering department — there is only a single LED to indicate when compression is taking place, and no gain reduction meter. Furthermore, its optical compressor is designed to impart a definite character to the sound, but we wanted to hear what we were getting as honestly as possible, so that we could determine how much effect the tape machine was having on the sound.

The result this time was more natural, but it became clear the tape machine was losing some of the high-end detail. Nick was also reusing old tapes acquired with the machine, and it was unclear what type of tape the machine had been set up f

Although EQ'ing the vocal with Nick's Aphex 109 produced very good results, Paul also tried out the TL Audio Ivory equaliser with similar settings, and Nick found the more 'glassy' sound more to his taste.
or, or indeed when it had been last serviced. Although Nick keeps the heads clean, there was evidence of head wear too, which would also tend to diminish the higher frequencies. So, we added 4dB at the high end using Channel One's Air EQ control (which is simply a wide band-pass boost at around 14kHz) in an attempt to compensate and kept the compression fairly gentle with a gain reduction of no more than 5dB on the loudest peaks. At this point we had something coming back off tape that our ears told us was generally similar to the original performance.

Nick still wanted more presence, so we patched in his Aphex 109 and treated the vocal track to 3dB of 7kHz boost with a 1.5-octave bandwidth and also pulled down the 180Hz region by 1.5dB to clarify the low end. This was definitely closer to the contemporary sound Nick was after and, because his vocal performance is dynamically well controlled, no further compression was felt to be necessary. Nick's singing voice was also free of sibilance, so adding to the top end was vice-free! Interestingly, we tried a similar EQ treatment using the TLA units and found that the sound was quite different and in some ways more 'glassy'. Nick rather liked the effect and resolved to experiment further with the different EQs.

Working On Acoustic Guitar

Though Nick hadn't specifically asked about the acoustic guitar sound, I was curious as to how he recorded it, and he told us that, for his Takamine, he used a combination of mic and DI with the mic (his TLM103 again) pointing at the junction of the neck and body. Hugh observed that combining mic and DI can sometimes lead to phase problems that colour the sound, so we suggested that Nick try recording with just the microphone. Nick commented that the miked sound was often insufficiently bright (hence the need to add the DI) so we offered to help hi

Moving the recording mic while listening to the results on headphones allowed Paul to find a better guitar sound.
m find a new mic position that would give him the kind of result he wanted without having to add in any of the DI sound.

The easy way to find the best mic position is to wear headphones and listen to the mic output as you move it around the guitar, so Nick played and I adjusted the mic. Sure enough, the usual neck-meets-body position was slightly lacking definition so I moved the mic further up the neck. One very interesting result was achieved by miking directly over the headstock from around six inches away — very bright and ringy — but the most natural sound we got was miking just a few inches above the third fret.

Acoustic guitars on pop records are often EQ'd to remove the low end and they may also be EQ'd to make them sound brighter than they really are, so we patched in the Aphex 109 equaliser. What we came up with was 5dB of shelving cut at 40Hz combined with 5dB of boost at 15kHz, with the bandwidth control set to its widest position. This produced the characteristic zingy, bottom-light sound that sits so well in pop mixes. A little gentle compression (around 6dB) and a hint of reverb completed the picture. Because we'd eliminated the DI component, the resulting sound was more open and natural than before, and that made the guitar sit well without the bottom end clouding other elements of the mix.

Setting Up A Better Reverb

Next we turned our attention to the reverb treatment. Nick had been using a Hall setting, but felt that something dr

ier and brighter might work better. After a little experimentation with the PCM90 presets, we came up with the Club/Air patch and turned the Club part of the sound down slightly using the Adjust knob (which is normally mapped to this parameter). Right away the sound became better focused and more upfront, and in direct comparison with the original vocal track Nick had recorded using his earlier methodology, the improvement was significant.

Nick had been worried that our recommendations would involve him spending a lot of money, but now it seemed as though a more transparent mic preamp, ideally one with the ability to apply EQ boost in the 14kHz region, and some extra acoustic treatment might do the trick. Hugh also suggested that Nick should invest in a shockmount for the TLM103, partly because the standard mic clip was broken anyway and this lovely (and expensive) mic was in danger of being dropped on the floor! Hugh also suggested that it would be a good idea to put a plastic bag over it when it wasn't being used to prevent dust getting onto the capsule. It is one of the drawbacks of electrostatic mics that they tend to attract dust if left out in the open, so either pack the mic away in the box — which can be a real pain — or protect it with a polythene bag over the top. A dusty diaphragm is also more prone to suffering from increased noise and frying sounds when the humidity rises.

Fixing The Mix

At this point we adjourned to the local pub for a splendid steak and ale pie, after which we made a start on the mix.

By using the separate outputs of his Roland R70 drum machine, Nick was able to EQ some extra warmth into the kick sound with his Aphex 109 compressor.
Prior to our visit, Nick had sent us a CD-R of some of his 'work in progress' and, aside from the vocal issues already addressed, Hugh and I both felt the kick drum and bass guitar were a little too far back in the mix. The kick drum was also very clicky and seemed to lack body. On quizzing Nick further about his mixing methods, we discovered that he normally works with an SPL Vitalizer Mk II across the mix buss, so we suggested that he bypassed it while he got the mix sounding as good as possible, and only then switch it in with a view to adding a subtle sparkle and to help overcome the limitations of his old Tascam 388.

To beef up the kick drum sound from the drum machine, we added 6dB of boost at 80Hz combined with around 4dB of cut between 250 and 300Hz — the latter prevented the bass boost from spilling over into the mid-range. The bass guitar needed more definition, so we applied exactly the same strategy as we did on our last Studio SOS and added a little boost at 350Hz to give it more of a 'miked amp' sound. A little judicious rebalancing and the track (which was essentially a guitar/bass/drums mix) started to gel nicely. The drum machine outputs were split into kick, snare and stereo kit, so we were able to add some plate reverb to the snare and kit using a Microverb. We used the Microverb's own EQ to add a little high-frequency boost, so there was some sizzle to the reverb sound. The kick drum was left almost dry — we added just enough reverb to get it to sit nicely with the rest of the kit. The other components of the mix were treated using the same PCM90 reverb patch that we used on the vocal track

Once the mix was sounding good, we returned to the Vitalizer and chose some more gentle settings than Nick had been using previously, with a view to enhancing the overall clarity and separation of the mix. This was a Mk II model that has a few more controls than the regular stereo model and, as I already own and use a Vitalizer, I made the necessary adjustments while Hugh and Nick listened. Firstly, I moved the Mid-Hi Tune control up from its 3.5kHz position to around 7kHz, so as to confine the enhancement effe

After working on the vocal sound with Paul's voice channel, Nick has decided to invest in a similar dedicated unit himself — probably a Focusrite Voicemaster in his case.
cts to the very top of the audio spectrum and to avoid adding harshness to the upper mid-range. The high EQ was set to 10kHz and the boost control set at around half way, while the bass end was treated to a little of the 'tight' bass boost. Vitalizer users will know that the bass end is addressed via a single control that offers a deep, warm bass when turned anticlockwise from its centre position and a tighter, more focused bass when turned clockwise. The Process level control sets the overall degree of processing, and this was reduced to a setting just below half-way. Switching the Vitalizer in and out of circuit showed that what we were getting now was a more subtle effect that retained the tonal feel of the original more than before, but still aided the clarity and separation of the various elements in the mix. It may be that little or no Vitalizer processing would have been necessary had Nick been using a digital recorder, but in the case of the 388 it definitely helped.

At this point Nick asked what improvements could be achieved using good mastering equipment. We explained t

  Improvising A Temporary Repair For A Damaged Speaker  
  For monitoring, Nick had a pair of Tannoy Reveal passive monitors fed fro
A brief application of the vacuum cleaner provided a quick fix for a damaged soft-dome tweeter — however, you can still see the creases in the reshaped tweeter, and a replacement would be required in the long term.
m a Samson power amplifier, though he'd managed to push in one of the soft-dome tweeters in getting the speakers from home to the studio. This didn't sound as bad as you might have expected it to, but I thought it was worth trying to fix the tweeter by using Nick's vacuum cleaner to pull the dome back into shape. By gently placing the end of the hose over the tweeter with the cleaner running, the worst of the damage was reversed — but before you try this at home, I'm offering no guarantees that this method will work for you!

Although pulling the tweeter dome back into shape in this way will often appear to repair the damage, any damage caused by the creasing remains and increases the distortion at high frequencies considerably above that of an undamaged tweeter. The best solution by far is to replace the tweeter, and if the speakers are more than a couple of years old or have had a hard life, it's best to change the tweeters in both speakers. For critical work, tweeters should be replaced every five years or so anyway, as they deteriorate over time.

 
hat using multi-band compression, top-quality EQ and separate limiting allows the mastering engineer to fine-tune the mix much more effectively than conventional full-band processors. Also, mastering houses invariably have more accurate monitoring systems than project studios, so you get to hear what the recording really sounds like. And of course there's the experience of the guy doing the work and the fact that it is a fresh pair of ears listening to the mix. A good mastering engineer hears what is really there, not what the musician thinks is there! The full details of mastering are too complex to go into here, but we did stress that if he was considering getting any of his mixes professionally mastered, he should leave out any global processing, such as compression or Vitalizer, altogether and let the mastering engineer do the processing his own way.

Homeward Bound

What we learned from this visit was once again that problems usually have several causes. The room problems were easily fixed using extra acoustic treatment and by working closer to the microphone, but the use of an aging tape machine with Dbx noise reduction was clearly a major factor in the sound. To get the vocal sound we wanted involved not only sorting out the acoustic problems but also using a more transparent preamp, adjusting the recording levels, using a good quality external equaliser and finding a more sympathetic reverb treatment.

We also discovered that the 388's own mixer section EQ sounded quite harsh and grainy compared to the outboard EQ Nick had bought. Having a Vitalizer turned out to be fortuitous, as it provides a fairly simple way to add 'fairy dust' to final mixes and it also helps compensate for the high-end loss incurred by the tape machine.

  Session Notes: Nick's Views  
  "Having been very dissatisfied with the vocal recordings I was making, I have decided to upgrade my mic pre, probably to the Focusrite Voicemaster, and will ensure that, when I record with a mic in future, I rig up some damping akin to the duvet which Paul valiantly held up while recording the test tracks. It made the sound markedly less boxy. It was also helpful to see the EQ settings which Paul and Hugh came up with for both vocals and acoustic guitar on my parametric EQs.

One thing that surprised me was how good my acoustic sounded when miked up about four inches from the third fret. I've always found my Takamine difficult to record, but had never tried this mic placement before. It brought out the top end of the guitar without making it sound harsh, whilst avoiding the boomy sound I was getting placing the mic nearer the body/soundhole of the instrument. Paul and Hugh spent some time trying different reverb presets on my Lexicon PCM90, and settled on one which avoided an obvious reverb tail, whilst lending the sound sufficient ambience to allow it to sit in the mix.

When Paul expressed the view that the Dbx noise reduction on my Tascam 388 was a fairly unsubtle system, and that it might well be having an adverse effect on my recordings, it confirmed my own suspicions, and it has hardened my resolve to acquire a stand-alone hard disk recorder. Hugh also advised me to invest in a balanced patchbay to get the best out of the outboard that I presently wire through an unbalanced patchbay. I see the sense in that and, funds allowing, will try to implement that upgrade at some point.

I have been recording at home for some time without any formal training or guidance, and had become very uncertain as to what I was doing right and wrong. It was extremely useful to have some reassurance that there was nothing I was doing that was obviously daft. I was left with the view that I was heading in the right direction, which was supportive. The tweaks which Paul and Hugh made had a significant impact on the mix, making it seem more balanced and professional.

The opportunity to speak to two people with such experience was invaluable, and has left me with a much clearer idea of what I can expect from the equipment I have, and where its limitations lie. Many thanks to Paul and Hugh — it really was a fantastic experience to have my own personal SOS clinic!"

 

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