More Of Everything!

Maximising The Loudness Of Your Masters

Published in SOS March 2003
Technique : Mastering

We provide some tips for making your

masters sound as loud as commercial
tracks, without sacrificing too much sonic
quality along the way.

Paul White

I've attended more than one discussion with eminent professional

mastering engineers, and the quest for 'loudness at all costs' is
one of their main concerns, specifically because the processing
involved in achieving this is often detrimental to other aspects of
the music. On the other hand, they are pragmatic enough to
acknowledge that their work has to compare in loudness with other commercial releases, otherwise there will be a
noticeable level disparity when their records are played alongside others. For that reason, rather than simply providing a
'one size fits all' recipe for loudness maximisation, I'm going to look at some of the strategies you can apply and, where
applicable, discuss the detrimental side-effects that might be experienced. But first of all, what is loudness in the context
of a digital recording?

Digital Loudness

As with analogue recording systems, digital recorders can 'under-record' a signal if the peaks in the music being
recorded aren't allowed to peak at digital full scale, so making proper use of the available headroom is clearly a basic
concern. In reality you need to leave a few decibels of safety margin, but try not to leave more than you have to. Most
digital workstations include a process known as normalising, which searches the file for the loudest peak, then increases
the gain of the whole file so that the loudest peak just reaches digital full scale. Normalising can be used to bring up the
level of audio files that come from under-recorded sources, but, though the normalisation process restores the optimum
peak level, it doesn't recover any resolution lost by recording at too low a level. Furthermore, normalisation alone can't
guarantee that a track will sound loud, because our hearing systems respond more strongly to average signal levels than
to peaks. Note that normalisation shouldn't be applied to a file before you undertake whatever other processing is
needed, as every stage of processing will involve further level changes to some degree. If you must normalise, it is
usually better to do it later on in the process, but it may not be necessary to use it at all if one of the later processes
involves the type of mastering limiter that also allows the user to set the peak level value.

Many commercial CDs have been
heavily clipped, but this can be at the
expense of seriously compromised
sound quality.
To create a loud track, it follows that we need high average levels, but, as most of
you will know already, the highest peak levels in a typical mix come from percussive
sounds, such as drums, and from high-energy bass sounds. To get an artistically
pleasing balance, the other elements of the mix, such as pads, lead instruments and
vocals, may be mixed at a much lower level than these high-energy elements.
However, the first steps to loudness come well before you get to mastering — for
maximum power, you need to choose suitable drum and bass sounds that are
intrinsically 'loud sounding'. Perhaps the best way to illustrate what I mean by this is
to take the example of a synth bass line that uses a fairly pure sine wave as its tone
source. Played over a big speaker system, this may deliver an impressive sensation
of low end, but heard over smaller speakers the power may be all but lost, as such
speakers are incapable of reproducing very low frequencies with any degree of
efficiency. Assuming that we can't turn up the level of the bass sound any more, as
it's already near to digital full scale, what else can we do to make it louder?

We already know that our ears respond to average sound levels, so if you can lengthen a percussive or fast-decay synth
sound, it will be perceived as being louder than a very short version of the same waveform. A couple of brief experiments
will also confirm that, given notes of the same basic pitch, length and amplitude, those with more harmonics will always
sound louder than a pure tone. If you like the 'impression' of a pure bass tone, you can be subtle about adding harmonics
by processing it via a distortion plug-in, a tube simulator or even a guitar effects pedal. There are also dedicated
processors such as Waves' Maxxbass plug-in, which reduce the level of the fundamental and increase the level of higher
harmonics to achieve a bigger sound using less headroom. Intrinsic to the concept of a powerful bass sound is the fact
that our ears compensate for a missing or attenuated fundamental frequency by implying it from the harmonics that are
present. That means you can get a very powerful bass sound without it containing a huge amount (or indeed any!) of its
fundamental frequency, something that's exploited very neatly by the Maxxbass process. On the other hand, you may
simply want to explore your synths and sample libraries to see if there are any alternative sounds that are subjectively
'bigger' that still fit your musical ideas.

Drums needed to be treated slightly differently, because they always start with a percussive attack, but there are some
bass guitar and synth sounds that also have a 'spike' on the front, in which case some of the following observations still
apply. There's a limit to how much a drum beat can be extended in length to make it sound louder, but there are ways to
do it. Going back to the source, acoustic drums that use less damping produce longer notes than the same kit with heavy
damping, so it follows that the damped kit will sound quieter than the undamped one. It's also true that a drum sound
recorded in a live environment will sounder louder than one recorded in a dry room, because the early reflections
reinforce and lengthen the basic sound, even though the peak levels may be similar. If you don't have a suitable live
room to record drums, or if you're working with dry drum samples, an early reflection or ambience reverb program can be
used alongside careful compression to achieve a similar effect.

That leaves the 'spike at the front' element to deal with, because the higher the attack spike, the greater the dynamic
contrast between the peak level and the average level, which translates into less perceived loudness. Fortunately, there
are several strategies that can be employed here, the most obvious of which is to use a fast limiter to reduce the peak
level, which in turn will allow you to increase the overall level of the sound within a track with worrying about clipping. You
can also achieve a similar thing using a compressor plug-in set to a fast attack time, with a fast release of between 10
and 50ms. Using a high ratio (greater than 10:1) and setting the threshold to produce between 3dB and 6dB of gain
reduction, you should be able to use the compressor's make-up gain to push up the overall level by the same amount,
thus gaining several decibels more level and, hopefully, not compromising the sound in any noticeable way. This strategy
can be used on individual drum sounds, specifically kicks and big snare sounds, or you can apply it to a complete drum
loop. Be aware, however, that ready-made drum loops taken from sample CDs may already have been processed in this
way, so there will be a limit to how much gain you can 'claw back' before the sound quality suffers.

Another way to limit signal peaks is to use some kind of soft clipping device, such as a tube emulator or even a distortion
device on a moderate setting. All distortion-inducing plug-ins and effects reduce the peak levels of the signals being
processed, but rather than controlling the overall gain as compressors and limiters do, they work on individual cycles of
the waveform, which introduces harmonic distortion. Providing distortion is used in moderation, it can make drums sound
brighter, punchier and subjectively louder, while keeping the peaks under control. In some situations you might get a
better-sounding result using distortion instead of limiting, as the drop in peak level is partly compensated for by the fact
that the added distortion makes things seem louder. There's also no reason not to follow up the distortion with fast
limiting if you're really going for the biggest sound you can get. In either case, though, try not to overprocess the
individual sounds, as you'll be applying more processing when you get to the mastering stage.

Guerrilla Mastering

So far I've focused on issues you might need to address when composing and mixing your songs, but there's a lot you
can do at the mastering stage to make your mix sound bigger. Not all the things I'm
about to suggest are considered good mastering practice (especially if you use them all
at once!), but all can be and have been applied in pop and dance music production. The
simplest of all processes is the 'smile' EQ curve (so called because, on a graphic
equaliser, the faders form a curve a little like a smiling mouth), where the mid-range is
gently scooped by a few decibels. This approximates the way our ear/brain combination
perceives the same music when it is played louder, as our perceptual frequency curve
changes at different levels, causing low and high frequencies to predominate over the
mid-range at higher SPLs. Any reasonably flexible equaliser will help you achieve this,
and the most effective area to cut seems to be between around 200Hz and 1kHz.
Alternatively, you can use a product (such as the SPL Vitalizer) that creates a 'smile'
curve automatically using relatively few controls. In addition to making the music sound
more punchy, the smile curve can also enhance clarity and prevent mid-range muddle.

After (or before if you prefer the way it sounds) comes overall compression, which can
be single-band or multi-band. In terms of maximising loudness, multi-band compression
is probably best, as dynamic changes at one end of the spectrum don't influence
processing at the other end (specifically, changes in low-frequency level don't modulate Here a 'smile' curve has been
created using the channel equaliser
the higher audio frequencies), but because what we're going to do here is fairly subtle, it in Steinberg's Cubase SX.
doesn't matter too much if you only have a full-band compressor. The trick is to compact
the overall dynamic range in a fairly gentle way by setting a low compression ratio (between 1.1:1 and 1.2:1) then
adjusting the threshold until you get a gain reduction of three or four decibels. You'll need a fairly fast attack time for this,
and the shortest release time you can use without getting audible pumping, so start with a 5ms attack time and a 50ms
release time, and then adjust the release until you get the most natural sound. Don't worry that the 5mS attack time will
let some transients through, because the next stage is to use limiting to sort this out.
While you can use a compressor as a limiter when you're mixing, I've never found they work very well for mastering, so
use a dedicated limiter plug-in such as the Waves L1, the Emagic Adaptive Limiter or the TC Works mastering limiter, all
of which are arranged in such a way that the output can be normalised and limited at the same time. Hardware users will
find similar facilities in units like the TC Finalizer and the Drawmer Masterflow. As a rule, aim for no more than 6dB of
gain reduction, but listen carefully for side-effects, as in some cases you may be able to achieve more than this and in
some cases less before the sound quality suffers. The main things to listen for are gain pumping and a dulling of detail or
loss of clarity.

By this time your music should be significantly louder than it was originally, but you can go still further by using a tube or
tape emulator plug-in (such as Dsound's VL2 for TC Powercore) to apply soft saturation to the remaining peaks. This has
a somewhat different effect to applying the same treatment at the individual track stage, as what's being treated now is a
mixture of sounds and all will be affected together. Processes of this kind may be best applied before limiting, but as the
order of processing produces slightly different subjective results, you should experiment to see what works best for the
track you are currently mastering.

The final weapon in the guerrilla mastering arsenal is that of good old-fashioned
clipping. Conventional wisdom has it that digital clipping should never be allowed, as
it usually sounds awful, but it is equally true that short bursts of clipping can be
tolerated without doing the audio any noticeable damage, simply because the events
are too short for our ears to register as distortion. As a very general rule of even
more general thumb, periods of clipping should be kept to under one millisecond
(around 44 samples maximum at 44.1kHz sample rate) and if two or more periods of
clipping follow each other in quick succession then the maximum period of clipping
needs to be made shorter to prevent audible side effects. The only way to tell how
A good setting to start with when using much clipping you can get away with is to listen — and if you're lucky, your audio
multi-band compression for mastering. software will also warn you of clipping and tell you how many consecutive samples
were clipped. If you are recording acoustic music, then using clipping as a means of
squeezing a decibel or two extra gain out of it may not be a good idea, but when it comes to high-energy dance music,
clipping is frequently employed either at the mixing or mastering stage (possibly both!) and, as with most things in audio,
if you can't hear it working (or if it makes something sound subjectively better), then it's fair game.

A Final Word Of Caution

While I have described a few tactics that might make respectable mastering engineers cringe, I think they'd agree with
me when I say the most important part of the process is listening. You must treat your mixes with respect, part of which
involves using a good monitoring system. With all the processes described, it's a good idea to experiment by deliberately
overprocessing so that you can hear what side-effects to look out for. Also be aware that excessive processing can
actually detract from the subjective power of a piece of music, so don't process all the impact out of your mixes and don't
be guided by how flat the waveform levels look on screen. The only result that matters is the one that's heard, so use
these tools with care and restraint and you'll be surprised what you can achieve. You'll know you're successful when you
get all the level you want but your music still gives the impression of having plenty of dynamic range.
20 TIPS ON...
20 Tips On Home Mastering
Published in SOS February 1999
Technique : Mastering

There's a world of difference between what happens in a professional mastering suite and what the average project
studio owner can do at home. But as more computer-based mastering tools become available it's quite possible to
achieve very impressive results with relatively inexpensive equipment. Certainly there's a lot more to mastering than
simply compressing everything, though compression can play an important role. The most crucial tool is the ear of the
person doing the job, because successful mastering is all about treating every project individually. There's no standard
blanket treatment that you can apply to everything to make it sound more 'produced'.

Every mastering engineer has preferences regarding the best tools for the job, but if you're just getting started I'd
recommend a good parametric equaliser, a nice compressor/limiter, and perhaps an enhancer, such as an Aphex Exciter
or an SPL Vitalizer. You also need an accurate monitoring environment with speakers that have a reasonable bass
extension, and some form of computer editor that can handle stereo files. The latter should ideally have digital inputs and
outputs, though if you're using an external analogue processor you'll probably be going into the computer via its
analogue inputs, in which case these need to be of good quality too. A professional may want to start off with a 20- or 24-
bit master tape or to work from a half-inch analogue master, but in the home studio most recording is done to 16-bit DAT.
This shouldn't be a problem for most pop music, providing you proceed carefully.

Most mistakes are due to over-processing, and the old adage 'If it ain't broke, don't fix it' applies perfectly to mastering.
Don't feel that you have to process a piece of music just because you can -- you might find that your master sounds
worse than the original material. And now for the tips...

1. Where possible, handle fade-out endings in a computer

editor, rather than using a master tape that was faded while
mixing. Not only does the computer provide more control, it
will also fade out any background noise along with the music,
so that the songs end in perfect silence.

2. Editing on DAT is very imprecise, so when you beam the

material into the computer (digitally, if at all possible) clean up the
starts of songs using the Silence function. Use the waveform
display to make sure you silence right up to the start of the song
without clipping it. As a rule, endings should be faded out rather
than silenced, as most instruments end with a natural decay.
When the last note or beat has decayed to around 5% of its
maximum level, start your fade and make it around a second long.
You can also try this if the song already has a fade-out, though
you may want a slightly longer fade time. Listen carefully to make
sure you aren't shortening any long reverb tails or making an
existing fade sound unnatural.

3. Once you've decided on a running order for the tracks on the album, you'll need to match their levels. This
doesn't simply mean making everything the same level, as this will make any ballads seem inappropriately loud.
The vocals often give you the best idea of how well matched levels are across songs, but ultimately your ears
are the best judge. Use the computer's ability to access any part of the album at random to compare the
subjective levels of different songs, and pay particular attention to the levels of the songs either side of the one
you're working on. It's in the transition between one song and the next that bad level-matching shows up most.

4. If an album's tracks were recorded at different times or in different studios, they may not sit well together without
further processing. The use of a good parametric equaliser (hardware or software) will often improve matters. Listen to
the bass end of each song to see how that differs and use the EQ to try to even things out. For example, one song might
have all the bass energy bunched up at around 80 or 90Hz while another might have an extended deep bass that goes
right down to 40Hz or below. Rolling off the sub-bass and peaking up the 80Hz area slightly may bring the bass end back
into focus. Similarly, the track with bunched-up bass could be treated with a gentle 40Hz boost and a little cut at around
120Hz. Every equaliser behaves differently, so there are no universal figures -- you'll need to experiment.

At the mid and high end, use gentle boost between 6 and 15kHz to add air and presence to a mix, while cutting at 1-
3kHz to reduce harshness. Boxiness tends to occur between 150 and 400Hz. If you need to add top to a track that
doesn't have any, try a harmonic enhancer such as an Aphex Exciter -- high-end EQ boost will simply increase the hiss.
5. To make a track sound louder when it's already peaking close to digital full scale, use a digital limiter such as
the excellent Waves L1 plug-in or Logic Audio's built-in Energizer. In most cases you can increase the overall
level by 6dB or more before your ears notice that the peaks have been processed. A nice feature of the L1 is that
you can effectively limit and normalise in one operation. It's always good practice to normalise the loudest track
on an album to peak at around -0.5dB and then balance the others to that track, but if you're using the L1 to do
this, make normalising your last process, so that you can use the Waves proprietary noise-shaped dither
system to give the best possible dynamic range. Normalising or other level-matching changes should always be
the final procedure, as all EQ, dynamics and enhancement involves some degree of level change.
Proper re-dithering at the 16-bit level is also recommended if going direct via a digital output to the production
master tape, as it preserves the best dynamic range. Analogue outputs will be
re-dithered by the A-D converter of the recorder.

6. If a mix sounds middly or lacking in definition, the SPL Vitalizer can be very useful (even the very inexpensive Stereo
Jack version produces excellent results). This device combines EQ and enhancer principles in a single box, and one
characteristic of the Vitalizer process is that the mid-range tends to get cleaned up at the same time as the high end is
enhanced and deep bass is added. As with all enhancers, though, be very careful not to over-use it: keep switching the
process in and out, to preserve your sense of perspective. The same applies to EQ and dynamics -- check regularly
against the untreated version to ensure that you're not making things worse.

7. Have a CD player and reference material on hand to use as a comparison for your work. Not only does this act
as a reference for your ears, it also helps to iron out any inaccuracies in your monitoring system.

8. Overall compression can add energy to a mix and even out a performance, but it isn't mandatory. Music needs some
light and shade to provide dynamics. Often a compressor will change the apparent balance of a mix slightly, so you may
need to use it in combination with EQ. Placing EQ before the compressor results in any boosted frequencies being
compressed most, while placing it after the compressor allows you to equalise the compressed sound without affecting
the compressor operation. Which is best depends on the material being treated, so try both.

9. A split-band compressor or dynamic equaliser gives more scope for changing the spectral balance of a mix,
but these devices take a little practice before you feel you're controlling them and not vice versa!

10. One way to homogenise a mix that doesn't quite gel, or one that sounds too dry, is to add reverb to the entire mix.
This has to be done very carefully, as excess reverb can create a washy or cluttered impression, but I find Lexicon's
Ambience programs excellent for giving a mix a discreet sense of space and identity. If the reverb is cluttering up the
bass sounds, try rolling off the bass from the reverb send.

If you want to add a stereo width enhancing effect to a finished mix, there are two main things to consider: the balance of
the mix and the mono compatibility of the end result. Most width enhancers tend to increase the level of panned or stereo
sounds while suppressing centre sounds slightly. Sometimes this can be compensated for by EQ, but being aware of
what's happening is half the battle. Other than the simple phase-inversion width enhancement used in the SPL Vitalizer,
which is completely mono-compatible, width enhancement tends to compromise the sound of the mono mix, so always
check with the mono button in. While most serious listening equipment is stereo these days, many TVs and portable
radios are not, so mono compatibility is important.

11. Listen to the finished master all the way through, preferably using headphones, as these have the ability to
show up small glitches and noises that loudspeakers may mask. Digital clicks can occur in even the best
systems, though using good quality digital interconnects that are no longer than necessary helps to reduce the
risk.

12. Try to work from a 44.1kHz master tape if the end product will be a CD master. If you have to work from a 48kHz
tape or one with different tracks recorded at different sample rates, a stand-alone sample-rate converter can be used
during transfer of the material into a computer. If you don't have a sample-rate converter, most editing software will allow
you to do a conversion inside the computer, though this takes processing time and the quality is not always as good as
that from a good-quality dedicated unit.

When using a software sample-rate converter, ensure that the tracks are all recorded with the computer system set to
the same sample rate as the source material. If you don't have a sample-rate converter at all, don't worry too much, as
transferring in the analogue domain via decent external A-D and D-A converters may well produce better results than an
indifferent sample rate converter (with free re-dithering thrown in too!). Alternatively, if your master is for commercial
production rather than for making CD-Rs at home, leave your master at 48kHz and inform the mastering house so that
they can handle the conversion for you.

13. When you're transferring digital material into a computer, ensure that the computer hardware is set to
external digital sync during recording and internal sync during playback. Also double-check that your record
sample rate matches the source sample rate -- people will often present you with DAT tapes at the wrong sample
rate, or even with different tracks at different sample rates. All too often this is overlooked, until someone
realises that one of the songs is playing back around 10 percent slow!

14. Don't expect digital de-noising programs to work miracles -- even the best systems produce side-effects if you push
them too far. The simpler systems are effectively multi-band expanders, where the threshold of each band is set by first
analysing a section of noise from between tracks. For this reason it's best not to try to clean up your original masters
prior to editing, otherwise there may be no noise samples left to work from. With careful use you can achieve a few dB of
noise reduction before the side-effects set in -- as low-level signals open and close the expanders in the various bands,
the background noise is modulated in a way that can only be described as 'chirping'. The more noise reduction you try to
achieve, the worse the chirping, so it's best to use as little as you can get away with.

15. When editing individual tracks -- for example, when compiling a version from the best sections of several
recordings -- try to make butt joins just before or just after a drum beat, so that any discontinuities are masked
by the beat. However, if you have to use a crossfade edit to smooth over a transition, try to avoid including a
drum beat in the crossfade zone, or you may hear a phasing or flamming effect where the two beats overlap. As
a rule, crossfades should be as short as you can get away with, to avoid a double-tracked effect during the fade
zone. As little as 10-30ms is enough to avoid producing a click.

16. On important projects, make two copies of the final mastered DAT (one as a backup) and mark these as Production
Master and Clone. Write the sample rate on the box, along with all other relevant data. If you include test tones,
document their level and include a list of all the track start times and running lengths for the benefit of the CD
manufacturer. As mentioned earlier, if, for any reason, you have produced a 48kHz sample rate master, mark this clearly
on the Production DAT Master so that the CD manufacturer can
sample-rate convert it for you.

It's always a good idea to avoid recording audio during the first
minute or so of a new DAT tape, to avoid the large number of
dropouts commonly caused by the leader clip in the tape-spool hub. You can, however, use this section to record test
tones, which will also demonstrate to the person playing your tape that it isn't blank! If you put DAT start IDs on each
track, check them carefully to make sure that there are no spurious ones, and don't use skip IDs.

17. When deciding on how much space to leave between tracks on an album, listen to how the first track ends
and the second one starts. Gaps are rarely shorter than two seconds, but if the starts and ends are very abrupt
you may need to leave up to four seconds between tracks. Use the pre-roll feature of your digital editor to listen
to the transition, so that you can get a feel for when the next track should start.

18. When using a CD-R recorder to produce a master that will itself be used for commercial CD production, ensure that
the disc can be written in disc-at-once mode rather than a track at a time, and that your software supports PQ coding to
Red Book standard. Check with your CD manufacturer to confirm that they can work from CD-R as a master, and take
note of any special requirements they may have. Be very careful when handling blank CD-Rs -- there are commercial
CDs on the market with beautiful fingerprints embedded in the digital data!

19. Be aware that stand-alone audio CD recorders usually have an automatic shut-off function if gaps in the
audio exceed a preset number of seconds, usually between six and 20. This may be a problem if you need large
gaps between tracks. Occasionally, even very low-level passages in classical music can be interpreted as gaps.
Also note that these recorders will continue recording for that same preset number of seconds after the last
track, so you'll need to stop recording manually if you don't want a chunk of silence at the end of the album.

20. When making a digital transfer from a DAT recorder to a CD recorder that can read DAT IDs, it's best to manually
edit the DAT IDs first, so that they occur around half a second before the start of the track. Then you don't risk missing
part of the first note when the track is accessed on a regular CD player. Alternatively, there are commercial interface
units (or CD-R recorders with the facility built in) that delay the audio stream in order to make coincident or slightly late
DAT IDs appear before the audio on the CD-R.