Library of recording tips
The microphone preamp is possibly one of the most important component in the whole recording chain apart from the microphone itself. Between them, these two units bring the minuscule voltage coming from the diaphragm up to the 1+volts that we work with. From that point on everything is operating at normal levels and the signal can be recorded, EQ’d etc. without problems with noise and distortion. The typical mike preamp can have the following controls.
I said “can have” because not all do. First there is the mike input level knob. This is the gain control for the preamp.
A special note here – Before you start to set the mike input level you must set the console up for unity gain. This involves first setting the console output faders to Zero, then the channel fader to zero. If you are going out a group put that fader to zero. This first step is vitally important because a console is capable of increased noise and distortion if not setup with correct gain structures. If you have a little Mackie or something which doesn’t have a separate control over monitoring turn your amp and speakers down. Basically if you run the output faders low you have to get the gain from somewhere so you turn up the mike preamp which is capable of adding noise and distortion.
If you find that you are fully counterclockwise and still have too much signal you must insert the Pad. The pad drops the level by 10 – 20db (depending in the console) and stops the preamp from overloading. Some consoles will include a phase reversal button which is a very handy option to have. There is also a button for Phantom Power which will supply power to your mikes if required. (See microphones) It often comes as a single on/off switch on the rear of the console. There is often an optional line input knob with an associated mike/line switch. This allows you to trim the level of the line inputs. Finally you will probably find a mike/line button that allows you to adjust the level of the line input individually. The flip button is only on certain consoles. It swaps the two main faders over (line and monitor), More about this option later when we look at monitoring..
Before we look at compressors and limiters we must understand the term Dynamic Range. The Dynamic Range of a sound is the range between the quietest section and it’s loudest section or in the case of a recorder the range between the noise floor and the point of distortion. You know how loud a Symphony Orchestra can get yet you also know how quiet it can get. An Orchestra has a wide Dynamic Range.
The meters above show a dynamic range of 72db. On a home cassette recorder the quiet section in this track would be below the noise of the tape recorder and all you would hear through the quiet passage would be tape hiss. The distance from the loudest section to the point of distortion is called the Headroom. If distortion is reached at +6db then we currently have a 4db headroom. To reduce the dynamic range you could ride the whole track with a fader and turn it up when it’s too low and pull it back when too high or your can use a compressor.
In the diagram above unity gain means that what you put in you get out. In the 2:1 ratio example When the signal is above the threshold the signal output is reduced in a ratio of 2db in will give 1db out when the compression ratio is set at 2:1, so you have saved 1 db off the top of your dynamic range and you can turn it all up by 1 db without effecting the headroom. In a more severe case like the 20:1, which is more commonly called limiting, for a 20 db rise in signal only 1 db comes out. The compressor and limiter can be used together in one unit where the compressor works in the 2 – 15:1 range whilst the limiter stops the extreme transient peaks in the signal in the 15 – 20:1 ratios which is why it is often called a Peak Limiter.
In the above graph the threshold of the limiter has been raised so that the main program material will be compressed above the threshold of compression at 2:1 and above the limiting threshold it will be 20:1. A compressor is a gain reduction device, therefore all compressors have a make up gain control so that if you are using 3db of gain reduction you can turn the output by that amount and still retain the same headroom.
In the diagram above the transition from unity gain to compression at the threshold of compression/limiting is gradual instead of a straight line. This is called a Soft Knee threshold and is much smoother.
The Meter on a compressor can usually be switched to read either the input level, output level or the amount of gain reduction. It is advisable to check that the input level is correct before you start adjusting the threshold and setting the compression ratio etc.
The attack time determines how quickly the the compressor reacts to signals above the threshold. Signals have short sharp peaks called Transients that can easily trigger a compressor to act. The attack time determines how long the peak should be above the threshold before compression takes place. These short transients are important in the clarity of a sound but don’t effect the loudness of the sound. The aim of compression is to make the instrument sound louder, to squeeze the dynamic range, therefore you may wish to lengthen the attack time and let the transients through (to be dealt with by a limiter if necessary) and the compressor will then be working on sustained levels above the threshold.
The release time determines how quickly the compressor lets go, or restores normal gain. If the release is too fast for the amount of gain reduction applied then the return to normal gain over and over as the signal moves above and below the threshold can cause what is known as pumping because the gain structure is changing rapidly. It is advisable to ask the player to play sustained notes and set the release so the change of gain is smooth. Instruments that have long sustaining notes like bass guitars should tend to use a slower release times than sharp percussive instruments like percussion. Most of the new generation compressors now have an Auto button that leaves it to the compressor to work it out, and they usually do it fine.
Take a look at a typical compressor and its controls:
The left section is the Noise Gate section. It has controls for the threshold at which the gate opens, the release time variable and a switchable fast/slow attack control. The centre section is the compression section with the standard controls over threshold, ratio, attack and release. The Peak/RMS switch determines how the compressor will track the signal i.e. its peak content or its RMS content. The Auto button is often an option where the compressor works out the attack and release times itself by analysing the program material. The hard/soft switch determines the Knee setting. The meter can read input or output levels plus it can read the amount of gain reduction. Finally there is the makeup gain control (Often just labelled output level) The link button is there if there are two compressors in the unit . Stereo Compressors have a link facility that makes one of the two compressors a master. (Usually the left compressor). All the controls on the master effect the slave compressor, so they both operate together. If the compressors weren’t linked any strong signal on the right would be gain reduced and the stereo image would move because centre panned instruments would vary in their left to right balance so when compressing a stereo signal make sure the compressors are linked.
The Electro and Warm options are computer additions not found in a stand alone analogue version. As you can see the threshold is below the peak signal so gain reduction is taking place as indicated in the attenuation meter. The ratio is set at 2.90:1 and there has been no make up gain applied.
The attack time is set to 3.66ms and the release is at 214ms and the control on them is manual (not auto).
The expander is a compressor in reverse. There are two types of expander. In some, signals above the threshold remain at unity gain whereas signals below the threshold are reduced in gain, whereas in others the signal above the threshold also has the gain increased. Therefore you can use an expander as a noise reduction unit. Set the threshold to be just below the level of the player when playing. When the player stops the signal will fall below this threshold and the signal is reduced in gain thus reducing the noise or spill.
The drawing above shows the different actions of compressors and expanders. The expander in the drawing is increasing gain above the threshold and reducing gain below the threshold.
Most recording in popular music today has had heavy compression. Recording are loud and in your face! As well as most of the components of a track being individually compressed the whole mix overall has been compressed and limited before going to CD. I don’t think that’s a bad thing.
A limiter is just a severe compressor where the compression ratios are high. On some units like the DBX 160 and the Aleisis compressors an additional Peak limiter control with a LED that flashes is supplied, but units like the Aphex Dominator are pure limiters and are very sophisticated in how they attack and control peaks and you can get some pretty hot “brick wall” mixes through them.
A De-esser is a frequency selective compressor/limiter that compresses only at a predetermined frequency. If set to the frequencies around the sibilance area of a vocal (4kHz – 8Khz, it varies between men and women,) the vocal will be compressed only at those frequencies which will reduce the sibilance. Sibilance is the peaks of high frequencies created by ‘S’, ‘T’, ‘C’s etc.
The new generation
The new generation compressors, expanders gates etc. in the new computer programs are worth a mention here. These compressors have one outstanding advantage over the stand alone compressor. They can read the signal ahead of time by extracting the signal from the hard disk ahead of time, analysing it and then outputting it in real-time. They know what is going to happen next which gives them a distinct advantage in maintaining smooth control over the signal.
The diagram above shows how a gate works on level. When the signal falls below the threshold the gate reduces the level to the specified reduction level. The attack time here determines how quickly the gate will open and the release time determines how fast it will close. Some gates have a Hold function that allows you to tell a gate to hold open for a set time once it is open and then the release time can take over and close the gate. This facility can stop the gate opening and closing quickly due to peaks. It can also be used as an effect, especially if it is put over the return from a reverb unit. If you have some reverb on say a snare, and you put a gate over the reverb return signal, you can get the hold function to hold the reverb open for a period set by the hold function and then to quickly close it by using a fast release. This effect is called Gated Reverb and is now a standard program in most reverb units.
A gate can also be set to be triggered by something else via a side chain. For example, if you put a gate over a room ambience mike you could use the snare mike to trigger it to open when the snare was hit and to close when the snare stopped. This is called Gated Ambience. Another effect is to put a hihat feel into the side chain and modulate the gate to open and close on a synth sound. The effect is a modulating synth with the attack and release times controlling the modulation.
Gates can also so linked so that one controls the other and when one opens the other opens as well. (Like the compressor) This is used in stereo gate situations like over stereo toms.
I got this chart off the web and it gives you an idea of how the different materials absorb sound at different frequencies.
Remember that full absorption is 1 whilst full reflection is 0
Absorption coefficients of common building materials and finishes Floor materials 125 Hz 250 Hz 500 Hz 1 kHz 2 kHz 4 kHz carpet 0.01 0.02 0.06 0.15 0.25 0.45 Concrete (unpainted, rough finish) 0.01 0.02 0.04 0.06 0.08 0.1 Concrete (sealed or painted) 0.01 0.01 0.02 0.02 0.02 0.02 Marble or glazed tile 0.01 0.01 0.01 0.01 0.02 0.02 Vinyl tile or linoleum on concrete 0.02 0.03 0.03 0.03 0.03 0.02 Wood parquet on concrete 0.04 0.04 0.07 0.06 0.06 0.07 Wood flooring on joists 0.15 0.11 0.1 0.07 0.06 0.07 Seating materials 125 Hz 250 Hz 500 Hz 1 kHz 2 kHz 4 kHz Benches (wooden, empty) 0.1 0.09 0.08 0.08 0.08 0.08 Benches (wooden, 2/3 occupied) 0.37 0.4 0.47 0.53 0.56 0.53 Benches (wooden, fully occupied) 0.5 0.56 0.66 0.76 0.8 0.76 Benches (cushioned seats and backs, 0.32 0.4 0.42 0.44 0.43 0.48 Benches (cushioned seats and b 0.44 0.56 0.65 0.72 0.72 0.67 Benches (cushioned seats and backs, fu 0.5 0.64 0.76 0.86 0.86 0.76 Theater seats (wood, empty) 0.03 0.04 0.05 0.07 0.08 0.08 Theater seats (wood, 2/3 occupied) 0.34 0.21 0.28 0.53 0.56 0.53 Theater seats (wood, fully occupied) 0.5 0.3 0.4 0.76 0.8 0.76 Seats (fabric-upholsterd, empty) 0.49 0.66 0.8 0.88 0.82 0.7 Seats (fabric-upholsterd, fully occupie 0.6 0.74 0.88 0.96 0.93 0.85 Reflective wall materials 125 Hz 250 Hz 500 Hz 1 kHz 2 kHz 4 kHz Brick (natural) 0.03 0.03 0.03 0.04 0.05 0.07 Brick (painted) 0.01 0.01 0.02 0.02 0.02 0.03 Concrete block (coarse) 0.36 0.44 0.31 0.29 0.39 0.25 Concrete block (painted) 0.1 0.05 0.06 0.07 0.09 0.08 Concrete (poured, rough finish, unpai 0.01 0.02 0.04 0.06 0.08 0.1 Doors (solid wood panels) 0.1 0.07 0.05 0.04 0.04 0.04 Glass (1/4" plate, large pane) 0.18 0.06 0.04 0.03 0.02 0.02 Glass (small pane) 0.04 0.04 0.03 0.03 0.02 0.02 Plasterboard (12mm (1/2") paneling on 0.29 0.1 0.06 0.05 0.04 0.04 Plaster (gypsum or lime, on masonry) 0.01 0.02 0.02 0.03 0.04 0.05 Plaster (gypsum or lime, on wood lath) 0.14 0.1 0.06 0.05 0.04 0.04 Plywood (3mm(1/8") paneling over 31.7mm 0.15 0.25 0.12 0.08 0.08 0.08 Plywood (3mm(1/8") paneling over 57.1mm 0.28 0.2 0.1 0.1 0.08 0.08 Plywood (5mm(3/16") paneling over 50mm( 0.38 0.24 0.17 0.1 0.08 0.05 Plywood (5mm(3/16") panel, 25mm(1") fi 0.42 0.36 0.19 0.1 0.08 0.05 Plywood (6mm(1/4") paneling, airspace, 0.3 0.25 0.15 0.1 0.1 0.1 Plywood (10mm(3/8") paneling, airspace 0.28 0.22 0.17 0.09 0.1 0.11 Plywood (19mm(3/4") paneling, airspace 0.2 0.18 0.15 0.12 0.1 0.1 Absorptive wall materials 125 Hz 250 Hz 500 Hz 1 kHz 2 kHz 4 kHz Drapery (10 oz/yd2, 340 g/m2, flat aga 0.04 0.05 0.11 0.18 0.3 0.35 Drapery (14 oz/yd2, 476 g/m2, flat agai 0.05 0.07 0.13 0.22 0.32 0.35 Drapery (18 oz/yd2, 612 g/m2, flat agai 0.05 0.12 0.35 0.48 0.38 0.36 Drapery (14 oz/yd2, 476 g/m2, pleated 0.07 0.31 0.49 0.75 0.7 0.6 Drapery (18 oz/yd2, 612 g/m2, pleated 0.14 0.35 0.53 0.75 0.7 0.6
Fiberglass board (25mm(1") thick) 0.06 0.2 0.65 0.9 0.95 0.98 Fiberglass board (50mm(2") thick) 0.18 0.76 0.99 0.99 0.99 0.99 Fiberglass board (75mm(3") thick) 0.53 0.99 0.99 0.99 0.99 0.99 Fiberglass board (100mm(4") thick) 0.99 0.99 0.99 0.99 0.99 0.97 Open brick pattern over 75mm(3") fiber 0.4 0.65 0.85 0.75 0.65 0.6 Pageboard over 25mm(1") fiberglass bo 0.08 0.32 0.99 0.76 0.34 0.12 Pageboard over 50mm(2") fiberglass bo 0.26 0.97 0.99 0.66 0.34 0.14 Pageboard over 75mm(3") fiberglass b 0.49 0.99 0.99 0.69 0.37 0.15 Performated metal (13% open, over 50mm 0.25 0.64 0.99 0.97 0.88 0.92 Ceiling material 125 Hz 250 Hz 500 Hz 1 kHz 2 kHz 4 kHz Plasterboard (12mm(1/2") in suspended 0.15 0.11 0.04 0.04 0.07 0.08 Underlay in perforated metal panels (25 0.51 0.78 0.57 0.77 0.9 0.79 Metal deck (perforated channels,25mm(1" 0.19 0.69 0.99 0.88 0.52 0.27 Metal deck (perforated channels, 75mm(3 0.73 0.99 0.99 0.89 0.52 0.31 Plaster (gypsum or lime, on masonary) 0.01 0.02 0.02 0.03 0.04 0.05 Plaster (gypsum or lime, rough finish 0.14 0.1 0.06 0.05 0.04 0.04 Sprayed cellulose fiber (16mm(5/8") on 0.05 0.16 0.44 0.79 0.9 0.91 Sprayed cellulose fiber (25mm(1") on s 0.08 0.29 0.75 0.98 0.93 0.76 Sprayed cellulose fiber (25mm(1") on 0.47 0.9 1.1 1.03 1.05 1.03 Sprayed cellulose fiber (32mm(1-1/4") 0.1 0.3 0.73 0.92 0.98 0.98 Sprayed cellulose fiber (75mm(3") on s 0.7 0.95 1 0.85 0.85 0.9 Wood tongue-and-groove roof decking 0.24 0.19 0.14 0.08 0.13 0.1 Miscellaneous surface material 125 Hz 250 Hz 500 Hz 1 kHz 2 kHz 4 kHz People-adults (per 1/10 person) 0.25 0.35 0.42 0.46 0.5 0.5 People-high school students (per 1/10 0.22 0.3 0.38 0.42 0.45 0.45 People-elementary students (per 1/10 p 0.18 0.23 0.28 0.32 0.35 0.35 Ventilating grilles 0.3 0.4 0.5 0.5 0.5 0.4 Water or ice surface 0.008 0.008 0.013 0.015 0.02 0.025
RT60 relates to intelligibility. Diffractors reduce pronounced reflection by breaking up the sound wave before reflecting it back. This does not reduce reverberant energy, but does reduce echo spikes that may otherwise exceed -60db of direct, thus lowering RT60 and improving intelligibilty, but not necessarily improving the listening environment for music.
Low frequencies are big waves, consider that a 50Hz wave is 6.6m (21′ 8”) and a 30Hz wave is 11m (36ft) long! That’s 11m peak to peak -There’s a lot of guys around here who would love to surf a wave like that! So to stop it requires special techniques.
There are basically two ways to control low frequencies.
- Acoustic Hangers. This is a system of fibre board panels that are wrapped with insulation and are hung freely using wire or rope. The large hangers 1.8m x 500mm work in the low frequency range whilst the panels 1.2m x 300mm effect the low mid frequencies. It is common to have up to a 1.2m space at the rear of the control room with the large hangers whilst the smaller hangers are effective if suspended in the ceiling cavity created by a false ceiling.
- Panel Absorbers. A panel of plywood or particle board is placed over an air cavity with insulation glued to the back of the panel. The panel has a resonate frequency and when it occurs in the room it resonates and the insulation absorbs the energy.
The above drawing shows the rear of a typical control room design. The fibreboard panels are suspended from the ceiling with the sizes varying to give a broadband absorption field. They can also be hung behind a false wall in the studio as in the following drawing.
False Wall with Acoustic Hangers
A panel absorber is created when you place a sheet of plywood or fibreboard, with insulation glued to the back of it, over an air cavity. The panel will have a resonate frequency of its own, tap it and you will hear it. When it is placed over a sealed cavity, and insulation is attached to the back, everytime it hears its own note it resonates and the air in the cavity resonates and the insulation absorbs the resonance, hence absorbing the frequency! It is important to note that here we have an absorber that reflects the high frequencies and attenuates the low. With the hangers all that exposed insulation absorbs the high frequencies as well so the panel absorber has a place in the studio. The two factors determining the frequency of absorption are:
- The mass or density of the panel.
- The depth of the air cavity, i.e. depth of the sealed timber frame.
A panel absorber is made like this:
You can apply different shaped front panels
The other great advantage of panel absorbers is that they can have angled or curved fronts so when mounted on a wall or the ceiling they stop parallel wall interference and prevent standing waves creating diffusion.
You can even tune this absorber by placing a contact microphone on the plywood panel which is plugged into a real-time analyser and blasting the panel with white noise or a swept tone with a speaker. When the frequency = the panel’s resonate frequency the panel will vibrate and the frequency will show up on the real-time analyser. The thicker the plywood the lower the frequency and the greater the depth (depth of the timber box) from the wall the lower the frequency. Using fibreboard as an alternative tends to create a low-mid absorber.
You can create a broadband low frequency absorption wall by building a series of sealed boxes with different depths with each box being only 1m x 1m (3′ x 3′). With a variety of different thickness of plywood you can cover the whole low frequency range. It looks good too. You can also alternate the fronts between panels and slats. (See helmholtz resonators)
For absorption coefficients and panel thickness check out the absorption coefficient chart.
You can create a variable panel absorber by splitting the box into two boxes and placing hinges on one side so that it opens fully as per the following diagram:
VARIABLE PANEL ABSORBER
The variable panel absorber allows you to change the acoustics in a room. A wall of these absorbers can quickly change a room’s acoustics from live to dead. A variation is to have a slat resonator in the bottom box so that when the box is opened it reveals a slat resonator so you end up with a wall of alternating low-mid absorbers and high frequency absorbers. If you can only afford the space for one studio this is an excellent addition as you can change the room acoustically to cover all situations.
The scale along the bottom of the chart shows the frequencies from 16Hz to above 16kHz. When engineers talk about the high mids they are referring to the frequency range from 1kHz to 8Khz, roughly.
The centre frequency is around 750Hz and the Gain Increase (boost or cut) is around 18db. The Q Factor is the width of the frequencies effected by the boost and is measured in octaves. A high Q is narrow and a low Q is wide.
There are two kinds of equalisers, Parametric and Graphic and each can control a number of bands.
The Graphic Equaliser
You will note that there are slider controls for each frequency and the scale along the base shows which frequency. The scale along the top states how many db change has been made at each frequency and it can be positive or negative (boost or cut). A typical graphic equaliser does not have any controls over the Q factor of each boost, it is normally pre-set.
The Parametric Equaliser
The left unit is a typical high end console analogue equaliser whereas the right one is a new generation computer program digital ones. The left one has a switchable peak/shelf High frequency control. It has two sweepable mid bands with variable Q and a peak/shelf low frequency control. The computer version has 4 Bands each with it’s own centre frequency, Q width and gain. The resultant EQ curve is displayed as well. (It’s a digital EQ) The mid bands of the analogue version are usually divided into two sweepable bands the the low – mid covering 100Hz – 4Khz with the other covering 600Hz – 15Khz (typically – it varies from console to console) You will note that the digital unit is sweepable from 20Hz to 20KHz in all bands.
If you look at the coefficient of absorption figures for the various products you will note that whilst some attenuate the highs some also attenuate the low mids as well. 100mm (4″) fibreglass for example not only absorbs high frequencies but it also works down into the low mids depending on how thick it is.
The other main factor is what are the highs in your room doing? Consider the fact that your high frequencies are coming from your speakers which have a directivity factor. In a standard multi – speaker system the highs are coming from the tweeters or horns. Both these units have a fan shaped dispersion of around 30 degrees. And create what is referred to as the on axis off axis effect. Stand in front of a speaker and you hear all the highs but go 30 degrees off axis and the highs start to reduce to the point that if you are 90 degrees off axis the highs are eliminated completely (apart fro highs that reach you my reflection from some other surface.)
Take a look at this plan of a control room:
Control Room Plan
The dotted lines indicate the axis of the high frequency projection. Note that the engineer is sitting on axis to the speakers yet someone sitting to the right of the console is off axis to the right speaker but still on axis to the left speaker. The high frequencies are reflected by the opposing walls (in this case glass doors). The idea of this control room design is make sure (by angling the walls) that the high frequencies from the right speaker are not reflected back into your left ear.
Once the sound passes the engineer the rear of the control room absorbs the sound and it doesn’t come back to the engineer.
If you look at the absorption coefficients of various materials you will notice that some of the fibreglass products absorb low-mid frequencies very efficiently as does a panel absorber with a fibreboard panel instead of a plywood panel. But the best low mid absorber (and the best looking) is the helmholtz resonator – often called a slat resonator.
The helmholtz resonator (named after a Mr Helmholtz who discovered it) can best be demonstrated by taking a normal soft drink bottle and blowing over the mouth of the bottle – a note is produced. Now place some cotton wool in the bottle and try again. You will notice the note has reduced- well not really, the note is produced but the wool absorbs the resonance and turn the sound energy into heat! Imagine, if you lined a whole wall with bottles of various sizes, all filled with insulation material. You would now have a low-mid (200 – 500Hz depending on the bottle size) absorbing wall that as well as absorbing the low mids would also reflect or diffuse the high frequencies. I haven’t tried it yet but it would be worth trying if you are short of cash because bottles are cheap. The Romans used to do it using clay jars which they placed around their theatres.
The helmholtz resonator is often called a slat or slot resonator because you can create a helmholtz resonator by building a wall with slats of timber separated by slots as in the following diagram
|Further more, our scientists have created a formula with which we can tune the resonator to a specific frequency. If we vary the depth from the wall, slat width, slot width (and the slat depth) we can create a wall that is a broadband low-mid frequency absorber. The beautiful thing about these absorbers is that they still reflect high frequencies, in fact they will diffuse them which is even better.|
|As you can see a slat wall like this can break up parallel walls thus stopping standing waves. Because the distance from the front to the back is varying from 300mm to 100mm or around 12 degrees, the wall becomes a broadband absorber. So simple yet so effective! I’ve seen some beautiful looking ones where you cut the slots out of a sheet of quality particle board with a timber veneer.
Another form of helmholtz resonator is created using perforated plywood – i.e. plywood with hundreds of holes in it. We call it pegboard in Oz, you see it in hardware stores holding up tools etc. If you place a panel of this over an air cavity like in a panel absorber not only do the little holes act like bottle necks the whole panel acts as a low frequency panel absorber!
The formula for calculating the helmholtz resonant frequency is:
f = resonant frequency in Hertz (Hz)
Important thing to remember are:
Stereo room symmetry around your speakers.
- Glass windows or doors for communication.
Low-mid frequency absorption from 150 -550Hz.
High frequency absorption.
- Absorption across the rear of the control room wall.
- Whatever low frequency absorption you can fit in the space.
Things to avoid are:
- Having to go through the studio to get to the control room!! (I hate this because you always get interuptions as people move in and out of the studio)
- Creating studios with no visual communication. There is nothing worse than recording someone you can’t see.
- Big studios with a small pokey control room and visa versa.
So here are a few ideas that might start you off, use the selector for the different options.
THE BIG FACILITY
THE CORNER CONTROL ROOM
THE GARAGE STUDIO 1
THE GARAGE STUDIO 2
THE CONTROL ROOM
THE BEDROOM STUDIO
With the plethora of microphones around you’d be surprised at how engineers all over the world seem to use the same mikes. Go surfing to all the studios and you’ll find the same mikes in their mike list. I’ve got to state here that I’m not pushing any particular brand or type – I am not sponsored – so I’m only stating what I’ve observed over the years.
So how do they work? Basically all microphones have a diaphragm that vibrates when hit by sound waves. The vibration of the diaphragm is translated into an electrical signal that corresponds to the variation in the sound wave. That is why it is necessary to clean the diaphragms in your mikes on a regular basis as a build-up of dust, spit etc. will impede the vibration of the diaphragm and thus distort or colour the sound.
In a Dynamic microphone, also referred to as a moving coil microphone, the capsule is rather like a speaker in reverse. The cone is the diaphragm and it has a coil attached that is suspended in a magnetic field. When the diaphragm vibrates the coil creates an electrical current. This is an entirely passive circuit as the magnet can be a permanent one so no external power is required.
On the other hand the Condensor microphone has two plates, one fixed and one moveable, that are each charged with a polarising voltage that creates a capacitor. The vibration of the plates creates a change in the distance between them which changes the capacitance and thus the sound wave is converted into an electrical current. In this case external power is required as there is an electrical circuit required to produce the polarising voltage. Because the current obtained is so small an amplifier circuit is also included.
Thus when using Condensor mikes an external power supply is required. This can be either a stand alone power supply for one or more mikes or it can be fed to the microphone from the console down the microphone cable and is commonly referred to as Phantom Power and is now standard at 48 Volts and all new consoles have that facility and usually consists of an on and off switch on the rear of the console or is an on/off option on each module. Incidentally, don’t worry about sending phantom power to a dynamic microphone, it won’t blow it up as the circuit is inactive in a dynamic mic situation.
An Electret Microphone is also a Condensor microphone except that the charge on the plates is created by a permanent electrostatic charge. Therefore an external polarising voltage is not required but once again the voltage obtained is small so an amplifier is usually built in and powered by an internal battery. Electrets are often thought of as the cheap cousin to the condensor mike because the material required to hold the charge on the diaphragm is heavier but good electrets can sound fine.
The Pressure Zone
The PZM or Pressure Zone Microphone is also an Electret microphone except that it is mounted in a special housing near the pressure zone on the surface of a plate. This plate can be mounted on a flat surface like on the wall, floor or the lid of a piano. I have found that PZM mikes are not prone to popping and appear to have no proximity effect. They are typically used for pianos in concert situations where the lid can be closed to reduce spill and are also ideal as floor mikes in stage show productions.
The Ribbon Microphone consists of a thin metal ribbon that is placed in a magnetic field. The vibration of the ribbon within the magnetic field induces a current that is proportional to the variation in the sound wave. This is also a passive circuit as the magnetic field can be created by a permanent magnet.
The Valve Microphones
Finally I must say something here about Valve Microphones. As mentioned before, the signal from the diaphragm in a Condensor microphone is small and must be amplified before it reaches the console where again it is amplified further. It is within this area that signal deterioration can easily occur and therefore the quality of the microphone must also be judged by the quality of the first stages of amplification. In a valve microphone the Condensor stage is a standard condensor system but the amplifier section uses a valve circuit to amplify the current as opposed to a transistor circuit used in later models. When I first started as an engineer in 1966 all the Condensor microphones were valve and each had its own power supply. The introduction of the transistor microphone eliminated the need for power supplies because phantom power was invented for the purpose.
The other major factor in those days was signal to noise. The average tape recorder had a signal to noise ratio of around 58db as opposed to the 70+ with today’s analogue recorders.(Mainly due to the improvement in the surfacing of tape.) With such a low signal to noise ratio we were always careful about the high end of our recordings because if you had to add it later you sacrificed your noise and increased hiss. So when the transistor microphone came out we all remarked “Far out!” (it was the 60′s) listen to that top end!!” and immediately used them instead because records were getting brighter then. What we were hearing was the difference in distortion between a valve and a transistor. A valve distorts in the 2nd harmonic first whilst a transistor distorts in the 3rd harmonic. The 2nd harmonic distortion is smooth, we can handle it but 3rd harmonic distortion is hard and harsh to our ear hence the difference between the two. The valve appears warmer like a valve Marshall does compared with a transistor version.
Today, on the other hand, the top end and noise is not a problem as modern analogue tape recorders have good signal to noise ratios and our mike preamps are also quiet yet from another aspect it is. The top end of digital is extremely bright compared with analogue tape due to the inherent distortion of frequencies above 7kHz created by the slow sampling frequency of 44.1kHz which in reality produces close to a square wave above 10kHz I find it produces what I call digital fatigue. Rupert Neve was recently reported as saying that we will need to sample at 24 bit/192kHz to equal analogue. (We will eventually) Meanwhile the warmth of the valve acts with the harshness of digital and produces a great compromise, hence one of the reasons for the popularity of valve mikes today.
Alternatively engineers today will put a mike through a valve preamp which is the second stage of amplifying a mic signal. Once again it is the soft clipping of the high end that produces that warm sound. What a lot of manufactures do today is the put a valve within a transistor circuit thus obtaining the soft clipping of the valve with the improved signal to noise of the transistor circuits. I’ve even seen an ad for a CD player that has a valve circuit in it!!
A dynamic microphone has a set sensitivity pattern called Cardiod Pattern or “heart shaped” or “Kidney shaped” pattern and the response looks something like this.
NOTE: This is not the curve for a SM57!!
Please note that this is not the response curve of a SM57, a SM57 is might tighter than this, it is only a demonstration. The line through the centre of the mike is called the Axis and when standing directly in front you are said the be On Axis as opposed to being Off Axis at the side and rear. In this example at 0 degrees there is full sensitivity, at 90 degrees the signal is reduced – 5db, at 120 degrees by 10, at 150 degrees by 20 db etc.
Condensor microphones have the added advantage of being able to alter their pattern from the standard cardiod and produce either a Figure 8 pattern or an All-round pattern.
When using a Fig 8 mike you can place an instrument or singer on either side of the mike. With the all-round pattern you can place anyone anywhere as the pattern picks up through 360 degrees. Incidentally the all-round pattern does not exhibit proximity effect.
MS stereo is short for Mid Side miking. It is recognised as being the truest form of stereo miking because it is not subject to centre lift in mono. When you join a stereo signal into mono the instruments panned to the centre (i.e. equal left/right) lift in the balance and is referred to as Centre Lift. MS stereo recordings don’t have that tendency. You can buy MS Stereo microphones but if you’ve got a cardiod quality mike and condensor that will produce a Figure 8 pattern you are in business. Set them up like this.
The signal from the Figure 8 mike will need a mike splitter that splits into two mike inputs. This is the tricky part, To have a figure 8 mike it must be a condensor with phantom powering and if you split it and phase reverse, it will cut off the phantom power. You can purchase a transformer box like this:
The other way is to bring the Fig 8 mike up into a console and then take a feed from the direct out of that channel and bring it back in via a line input on another channel and phase reverse it.
Bring up the cardiod mike and pan it centre, now take the two splits of the figure 8 mike and pan one left and one right. Now reverse the phase of one of the splits. If you now have the cardiod mike pointing at what you are recording and you slowly add the fig 8 mikes you will hear the sound change from mono to wider and wider stereo as you add more of the fig 8 mike. The Cardiod mike is called the mid mike and the fig 8 is called the side mike.
When you mono this signal the left and right signal cancel each other and you are left with the mono centre signal which is a true mono. You can use MS Stereo for all sorts of things like overheads on a drum kit, ambience room mikes, stereo ACC guitars and pianos etc. You can always buy a MS Stereo mike but they are very expensive.
Anyone using microphones must understand proximity effect. When you get close to a microphone there is a rise in the low frequencies called the proximity effect. This low end boost can be 20+db boost at 50Hz!! A vocal mike like the Shure SM58 has a built in roll off to compensate for this because live performers like to sing close to the mike, but if you stand back from the mike it will start to sound thin, in other words if you want the SM58 to sound flat you must be close to it. Most mikes will have proximity effect so a low cut filter option is often supplied to compensate for it.
Microphone Phase Relationships
Before you record anything it is imperative that you check all your microphones for phase.
Two diaphragms in phase
Here the two diaphragms are moving in the same direction so they are in phase. Imagine them as two overhead mikes and they will both receive the signal from the drums in the same phase.
Two diaphragms out of phase
Here the two microphones are pointing opposite to each other yet their diaphragms are receiving the same signal. When the left mike’s diaphragm moves in the other mike’s diaphragm moves out.
As a result the two mikes are said to be out of phase and a phase reversal must be inserted or the two microphones will cancel each other. Officially they should cancel totally but they don’t entirely in practice because each has a slightly different signal because of it’s different position in the sound field. It will be most noticeable in the low frequencies so if you top and bottom mike a snare and don’t use a phase reversal the sound will be thin and lack low frequencies.
Similarly, miking toms top and bottom the bottom mike will require a phase reversal. If your console doesn’t have a phase reversal switch on it (funnily a lot don’t) you should build some phase reversal plugs of your own. This can be done by simply making a male to female mike lead with pins 2 and 3 reversed. It’s a good idea to paint them red or something so you know that they are phase reversal cables. You can also purchase pre-made phase reversal plugs from some retailers. Some people simply connect a male and a female cannon plug together with the leads reversed, paint them red and insert them into the mike lead before the mike patch point.
Checking your phase
A small note here – before you start recording it is a good idea to check the phase of all your microphones and cables. You can purchase small phase check boxes where you plug each end of the cable into it and if all three lights light up the cable is OK. At some stage it is worthwhile setting up all your mikes, select one mike as a reference, and getting someone in the studio to speak into your reference mike and each mike in turn to check that each mike is in the same phase and that all your cables are correct. You will notice immediately if one of your mikes is out of phase.
The Most Common Microphones
The famous D112 from AKG – a standard kick drum microphone.
The classic AKG 414 EB This is a great overhead – hihat mike (it’s also a great kick mike if you’re prepared to put such an expensive mike on the kick).
The AKG 451 is a beautiful all purpose quality Condensor microphone. It comes with various alternative diaphragm capsules with different pickup patterns.
The classic Sennheiser MD421 tom microphone. (John Laws has a gold one!! for you OZreaders)
Sennheiser MD441 is another great snare mike and can be used on toms.
The fantastic range of microphones from Neumann Germany. Unfortunately they are now so expensive that the average home studio owner can’t afford them. You can probably buy 10 SM57/58s for one budget Neumann!! But they do sound extremely good and are one of the best!! If you can afford at least one, preferably a pair, you’ll never regret it.
The classic Shure SM57. Probably the best value microphone available. You can use it on drums, guitars, vocals, whatever.
Without doubt the hardest part of recording is mixing, yet it is also the most enjoyable as this is when everything starts to come together and all the hard work justifies itself. A good mixer paints a picture in sound that attracts the listener and conveys the song clearly and simply. I could sum up a good recording as a series of priorities which are:
The “feel” or “groove”
The “fiddly bits”
The song is set from the start and a good producer will have chosen a song that has ‘something to say’ and a good mixer will convey that something to the listener.
The singer is the next most important aspect and a good mixer will allow the singer to be heard and the lyrics to be conveyed clearly but with style. There is nothing more annoying than hearing a track and not being able to distinguish the lyric amongst a babble of instrumentation. Fortunately you don’t hear recordings like that on commercial radio as they just don’t get a look in. Engineers are often guilty of cluttering up tracks with all sorts of tricks and garbage that distract from the song and the singer because they know the song so well after days in the studio that they think everyone hears it like they do. If the track is to have a chance of commercial success it must be understandable from the first hearing. Always underestimate the ability of the listener as they are not professional listeners like you.
The “feel” or “groove” is what catches the listeners attention initially and sets up the mood and emotion of the track. This is created by careful balancing of the rhythmic aspects of the track be it drums, percussion or a great guitar groove.
Finally there is the “fiddly bits” as I call them; they are the musical phrases linking lyrics, joining verses to choruses and filling solo sections etc. that are created by the guitar licks, the piano fills, the answering vocal phases etc.
So where to start?
Monitoring speakers come in two types. Nearfield and Main. I like to use both. I work primarily on the nearfield to establish my balances etc. and then every now and then I will switch it up to the big speakers as they give a better idea of the low frequency balance, plus it sounds good eh! (I was a Yamaha NS10 freak for years but now I’m totally sold on the Event 20/20. Well done Event!) To me a big speaker system is like a magnifying glass, it blows the sound up and you can hear more but for a big system to be really good you have to flush mount them and have good speakers and a good amplifier system. Can I say here that I don’t like equalised speaker systems. If they don’t sound good flat, get another speaker!!
The first important procedure is to setup your console for mixing. The first requirement is to setup your levels to and from your master recorder, usually a DAT. If your console has an oscillator send tone to the DAT and balance left and right channels. Then check that the return to your console, which is what you’ll monitor, is balanced correctly left and right. At this stage it is also recommended that you insert your master compressor either in the master stereo output inserts or inline between the console and the DAT and line up correct left/right balance. This procedure is very important as it effects your level structure from then on and if you don’t do it now you can end up with your levels all over the shop later.
Aux Sends and Returns
Next you must establish your auxiliary sends and returns.
One of the best ways to get perspective and separation within your mix is to what I refer to as “putting everyone in their own space”. You can achieve this through the use of reverb and effects. I like to have one reverb unit dedicated to the drums. No other instruments are sent to this effect, only the drums which will put them in their space. The choice of reverb for drums depends entirely on the track but I start by putting reverb on the snare and going through the presets to find the one that works best for the track. I find it usually ends up with a bright reverb of shortish length around 1 – 1.2sec reverb time.
Note: A very fine producer in OZ was once quoted as saying “Give me a studio with 10 Midiverbs over a studio with one Lexicon 224XL” We all know what fantastic units the Lexicons are but if it’s all you’ve got you are limited to only one perspective.
Next I’ll dedicate a reverb unit to act as my overall reverb effect. I look for the best (not necessarily most expensive) unit in the studio for this will be my master reverb for vocals etc.
In the example above there are 6 sends with 5 & 6 being an option over 3 & 4. I therefore like to use 1 for my drums and 2 for my master verb. Then I can assign the others for effects. I do this so that I can always add master reverb as well as effects if necessary and if I had used say 3, I couldn’t put master verb on channels where the effect was assigned to 5. Should I use a stereo or mono send to the effects?? To be perfectly honest I don’t think it matters. Most of the stereo input reverb units I find have a mock stereo input, not a true stereo. If you use two sends it really doesn’t make a difference unless you are working with the more expensive units like the aforementioned Lexicon, and even then I question the validity of two inputs especially if you are limited in the number of sends.
I then assign the sends 3 – 6 to additional effects like delay, pitch change etc. to act as perspective enhancers. When establishing delays I set them to the track tempo. See Tempo Chart. The idea is to add these perspective effects so you only just hear them when in solo and they appear to disappear when mixed into the track. Bob Clearmountain – the world famous mixer – always had two delays going, one on eighths and the other on 16ths. It puts an air around instruments and if mixed in correctly you won’t actually hear them, just sense them. Pitch change is another effect to consider with say the left channel set to -.008 cents and the right to +.008 cents. This effect is great on harmony vocals and it puts them in a different space form the lead vocal. Finally a soft flange or chorus is another effect I’ll have as an option for guitars etc. See Effects pages for settings.
Make sure that all your effects are returned through the effect returns and assigned to the master stereo output. If you are fortunate enough to have spare channels on your desk you can return your delay and chorus type effects back through a console channel as this gives you the option of adding master reverb to them and using the channel EQ. Delays can soften if master reverb is added to their returns plus you can attain your feedback from the console instead of using the control on the effect unit. Say you are using send 3 to a delay unit you can feed back to the delay by sending the send 3 on the return back into the unit. N.B. Incidentally, make sure that the dry/wet or mix controls on your effect units are set to wet as you are only wanting the effect from the units and you won’t need any dry sound. (If you are using the Alesis Quadraverb check this as all the default settings have 50% dry and 50% wet.) The returns from effects are usually panned full stereo L/R, but you may wish to bring the drum reverb back half L/R to separate the two.
Your console should now be setup like this
Some mixers start with the drums, others start with the vocal. I must admit I start with the drums as they convey the dynamic of a song. Hopefully you will have automation on your console, if not, you must now start setting up a series of moves and remember where and how they occur because, let’s face it, the balance within a mix is not static, it varies continuously throughout a song. For example lets say the drummer plays a rimshot snare through the verses and full snare in the chorus. The EQ required on the rimshot snare sound is probably different from the chorus snare sound so I often split the snare return from the recorder into two console channels so I can EQ and effect each separately and automate the switch between the two. For example, the snare in the chorus will probably require more reverb than the rimshot so having a separate channel allows for that. Automation also allows for the tom mikes to be muted when not needed thus reducing the spill of the rest of the kit and cutting out the constant ringing of the toms which occurs with undampened toms. The overhead mikes also will need to be ridden throughout the track, I tend to lower the overhead mikes when the rimshot is playing to achieve a tighter sound, then I lift them in the chorus when the full snare comes in. Reverb on the overheads gives reverb on the cymbals but it also adds reverb to the snare in the chorus and lifts the whole ambient sound of the kit. This has the effect of changing the perspective of the drums in a mix. You can also change the perspective by putting master reverb on the overheads which blends with the drum reverb.
Once we have achieved a reasonable balance of the kit and the dynamics are set in place we can add the bass. The bass and the kick drum will determine the bottom end of the track so the balance between the kick and bass is critical. The kick will give the bass punch and attack when they hit together.
Note: I must say a few words here about bottom end. The big mistake in mixing is to make the bottom end sound too big by adding lots of bottom end EQ to the kick and the bass. You must bear in mind how the track will be played back by the listener. Nowadays everyone has a stereo system with bass boost as an option either as a loudness switch or as a sub bass control. Everyone who has this option has it switched on!! If you get out a few of your favourite recordings and listen to them on your mixing speakers you will find that they are relatively shy in the bottom end and yet when played through your average boom box sound tight and fat. You have to start to understand what a flat response really means and learn to mix that way. If you put a bass on a VU meter you will notice how much energy there is in the bottom end. A bass peaking to zero will have the same apparent loudness as a highhat peaking to -30db. That’s because a hithat has no real bottom end compared with a bass so be careful with your low end EQ on basses and kick drums. I like to solo the two together and EQ them so that they are tight but not boomy.
Add the vocal
OK, so the bass and drums are now at their first mix level so next I will add the vocal and mix it sitting just above the bass and drums. This might mean an EQ change so they all sit tightly together. The vocal might need to be ridden with the automation and I’ll probably compress it again to keep the dynamic range within the boundaries of the whole track. I often find that the reverb on the vocal will need to be ridden so that the screaming high notes need more reverb that the quiet intimate sections in the verse. Here I take a feed from the direct out of the vocal channel and bring it up on another channel on the console. I then deselect this channel from the stereo mix output so it goes nowhere but the aux sends are still working. By adding reverb to this channel I can use this channel to ride the reverb on the vocal as an automated send.
Adding the rest
Now we can start to add the fiddly bits like the rhythm guitar and keyboard pads etc. adjusting their balance to fit tightly but not overpowering the vocal. (Please understand I am not defaming guitars etc. by calling them fiddley bits, they are just as important as every other part) The track should now be starting to take shape. If the dynamics of the drums and vocal have been set correctly the placement of the additional instruments will fall into place easily. The vocal harmonies, and solo instruments can now be mixed into the track and we are nearing the completion of the first mixdown.
Note: It is important to keep checking your mix in mono. Unfortunately stereo and mono are not compatible. When you switch to mono, instruments that are panned centre are 3db higher than in stereo so your vocal, kick and snare, for example, will come up in the mix. Some engineers actually make two mixes of a track: One that is full wide stereo with full dynamic range for home listening and one where all the hard left and right signals are panned to the centre or half centre and compressed for radio. It’s really hard because if you make a mix sound great on a good home hi-fi it won’t have the tightness and punch a mix made for commercial radio will have where the dynamic range is low. It’s common practice to make separate mixes of the singles from an album for radio whereas the remaining tracks are mixed totally for home hi-fi. I think you will find that most commercial records are mixed to sound great on FM Radio.
Rest and Recreation
It is important that you constantly give your ears a break during the mixing process as your ears have little compressors in them that will progressively shut your ears down. Have you noticed that when you’ve been in a loud club with a loud band when you go outside you can’t hear as well. It’s part of your ears protection system and a cup of coffee in another room watching TV or something will allow them to start opening back up. I like to “mix from the kitchen” as I call it. This means playing the automated mix and listening to it from an adjacent room with the control room door open, you’d be surprised how clearly you can hear the balance between instruments when you get away from the direct sound from your speakers. The relationship between the bass and kick, the balance within the harmonies, the clarity of the vocals etc. all become clearer when you relax and listen from another room.
Unfortunately the human ear is not flat at all levels. Some guys called Fletcher and Munson worked out what the response curve of the ear was and found that at low levels the ear missed out on the low frequencies and the high frequencies, whereas at loud levels it was the opposite.
From the above chart you can see that around 80 – 90db the ear is the flattest. The fact that we don’t hear low frequencies and high frequencies at low levels created the Loudness switch on stereo systems which boosts the low and high frequencies to compensate for the ear. Unfortunately, Joe Public doesn’t know this but knows that when it is switched in things sound fatter and brighter so they leave it in all the time. It is generally recognised that a level of 85db is where the ear is at it’s flattest so don’t mix too loud if you want a flat response.
The important thing about mixing is apparent loudness, or relative loudness. If I whisper into a mike and then I shout into a mike the shout will appear louder because I know that shouting is loud. It’s the same with mixing. You create an illusion of loudness, everything is relative. You can’t get bigger if you are already at your maximum. If I mix a soft acoustic guitar and vocal and peak to zero then bring in a full kit and grunge guitar also peaking to zero it will apparently get louder because I know that drums and guitar are loud. Mixing is the art of making signals that all peak to zero sound as if there is a dynamic range. Nowadays with the excellent compression systems we have most recordings are heavily compressed. I was told of a producer who hired a mixing engineer to mix an album. The guy turned up with racks and racks of compressors and set about compressing every track. He had one compressor for this and another for that etc. In the end the whole mix was pumping away and almost mixed itself. That album went on to sell millions of copies world wide. Those of you who have played with Waves Ultramaximiser will know what compression can do for a mix. If you watch most modern pop recordings on a VU meter the needle is almost static varying only a few db yet the tracks go from quiet intros to full on chorus and solo sections yet still there is only a small variation in level. So setting compression (and limiting) levels is important. I will always have a compressor across the output of my mixes as it helps control the peaks and brings up the loudness of the track but I may use individual compressors on separate channels.
Finally – do take the time to get a good mix. If you don’t you have not given justice to all the effort you put into recording it in the first place. It may take a few remixes, so what – it’s the final product that counts.