Daisy Grace Wakefield

22 October 2015

Better Mixes

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Daisy Grace Wakefield

22 October 2015

T4.2
Static Mix
A static mix is one which uses panning but does not use any
automation.
Out of the three mixes i created, automation was used on all mixes and
therefore i did not create a static mix. Reasons for this include;
On mix 1, within the football stadium, once the performers had sung

their football chant, they immediately started talking and creating

irrelevant material and therefore, using automation to slowly fade
them out was much more effective than just cutting out the
undesired audio. The fade tool was used in conjunction with this, but
automation was needed towards the end to help the audio fade out
faster.

For mix 2, Brotherhood war scene, it would have been a static mix if

the original audio was desired for the majority of the piece,

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beginning and middle, and faded out just before the end to have an
effect on the audience. As automation was used to achieve this, it
cannot be a static mix.
For mix three, Africa, automation was used to bring in the guitars

during the chorus as even with the use of compression and EQ, the
guitars took over the mix and other instruments such as vocals could
not be heard.

Automation

Signal to Noise Ratio (SNR)

SNR is a measurement which measures the level of desired

signal to that of unwanted signal or background noise. Sometimes
background noise is desirable, for example, if we were to mic up a
drum kit in a church, we would not place the microphones close to the
drums as possible like we would in a studio. There is a reason that the
drums are being recorded in a church and therefore we desire the
echo effect that the location will give when playing the drums. Other
times, the background noise is not desired as things such as breathing
and the noise from the air conditioning can be picked up and this is
what we do not want.
In my first mix, using EQ to remove unwanted frequencies of specific
instruments but cutting away them away, meant that any unwanted
background noise would have been removed from the mix. In this mix,
the alto saxophone kept picking up the kick drum, due to their
locations when recording was taking place. By cutting the active
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22 October 2015

frequency of
the kick drum,
roughly 30Hz
to 200 Hz , this
removed the
unwanted
noise. This
allowed me to
carry on with the
mix as the kick drum would have been overpowering.
Using automation is another effective way to remove unwanted noise.
Examples of background noise or disturbances could be heavy
breathing, scuffling of paper, squeaking chairs. These noises can be
removed this way without removing too much from the mix like if you
were to cut frequencies using EQ.
It would have been better if the instruments were recorded
individually as this would reduce the amount of unwanted background
noise and bleeding from other instruments.

dBFS and Headroom

dBFS stands for decibels relative to full scale in the digital world.
In a digital system, the highest value we can have without any
distortion or clipping occurring is 0 dBFS. As 0 dBFS is the highest
value in digital terms, all other numbers will be negative.
Headroom is the space between the signal level being received
and the maximum level the signal can be without distortion or clipping
occurring (0dBFS). Looking to the faders on logic, you want the
recording levels to stay in the yellow zone which is roughly inbetween -6dBFS and -12dBFS, which is classed a having good
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headroom on a mix. This also helps to avoid exceeding, or peaking

above, 0dBFS, distortion and clipping as these in the digital world are
not desirable.
To avoid very little headroom or no headroom at all within my second
mix, i found it was important to make sure that the levels being
received through the pre-amps, focus rite and apollo, while the
performers were practicing their piece, did not
Headroom
exceed 0dB on Logics faders. I used the gain

control on the apollo (vocals and keyboard) and the focus rite (drums)
to alter the signal levels. This would have caused problems for me
when i started mixing, as this would mean that it would have been
harder to create a mix without potential distortion.

Dynamic Range

The dynamic range is the difference in decibels (dB) between the

quietest and loudest point of the waveform.
Different genres of music have very different dynamic ranges for
example, if an orchestra was playing a classical piece, it will have
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obvious quiet and loud sections

to the piece as they would use
Image 1.1
the acoustics of the room they're
in to create their unique sound,
but with pop music, this level is
set at the same level throughout with very small dynamic changes.
Compression can alter the dynamic range.
For mix 1, the sine wave shows that there isn't any digital
distortion and that there is a good dynamic range of the audio file.

Lowest Points

Also for mix

2, there isnt
Highest
any
points
distortion. It
does seem to
peak at the
highest point
on the audio
file but there
is a small gap
between the
Distortion highest point and
where distortion
would occur.
For mix 3, its
clear that there
is a very high
level of
distortion
occurring within

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22 October 2015

the mix. Looking at the sine wave, it makes it easier to understand why
digital distortion is avoided as there will now be a very noticeable
fuzzy, raspy sound.

EQ / Frequency Spectrum

EQ can be parametric or semi-parametric. Parametric EQ means

that when we EQ an instrument, we are able to control the width,
decibels and hertz being cut or boosted and semi-parametirc EQ
means we can only manipulate decibels and hertz.
When we use EQ, we listen to the instrument selected and cut
away at the frequencies not associated with that particular instrument.
Frequencies associated with each individual instrument are called its
active frequency range. Different instruments have a different active
frequency range meaning that between certain hertz, the instrument
will produce a signal.
For example, a Pianos active frequency range starts at 30Hz and
going up to 4KHz, therefore we would cut below 30Hz and above
4KHz. Cutting unwanted frequencies allows more headroom within
your mix as this can get rid of any unwanted background noise (SNR).
Once unwanted frequencies have been cut away from
instruments, we can look at boosting frequencies. This means that
when listening to the
track, if there are any
instruments struggling
to be heard, we can
boost them by raising
the volume of that
specific instrument dB
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22 October 2015

at a time. By using a paramedic EQ tool, we are able to select the

frequency range, width, that is too quiet and increase its volume, dB.
Within my second mix, i used EQ to cut frequencies between 5kHz and
20KHz of the kick drum and boosted the lower frequencies of the
snare, left over head and
right overhead
microphones as the
cymbals and hi-hats
were very strong
Kick, Snare, Left overhead, Right overhead in the mix and
other parts of the
drum kit were buried.
As all instruments were playing together in the same room, there was
always going to be bleeding of instruments and by cutting away at the
dominant frequencies and boosting those lost in the mix after cutting,
this allowed instruments hidden to be heard.
For my first and second mixes, as the instruments were recorded in the
same room at the same time, the drums were miked up with one kick
drum microphone and overhead microphones in a spaced pair
manner. Despite only 4 microphones for the drums, they still bled into
others which meant carefully selecting the right frequencies of other
instruments to cut or boost to create a good balance within the mix.

Digital Vs Analogue Distortion

An analogue signal is a signal that does not contain breaks, in
other words is continuous. For example, our hearing is analogue as we
are hearing in real time, as the sounds reach us, and there are no
breaks, and analogue signal varies in amplitude and frequency.

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A digital signal contains breaks and can only have two values, 1
(on) and 0 (off). A digital signal is sent in a series of pulses containing
ones and zeros.
Digital distortion can result in a raspy and fuzzy sound. Digital
distortion occurs when the signals shape is altered in some way. This
can occur when a system such as an amplifier cannot cope with the full
signal and so cuts come of it off called clipping. This results in a
change within the dynamic range. Software will have some kind of
indictor which tells you its dBFS limit and as to where clipping starts to
happen.
When talking about analogue distortion, this is referring to the
character that the analogue processing/recording equipment and the
recording medium add to the sound [1]. http://
www.soundonsound.com/sos/feb10/articles/analoguewarmth.htm

With my first two mixes, during the recording session, i made

sure that there wasn't any digital distortion by adjusting the gain levels
on the pre amp before recording making sure that we did not peak
above
0dBFS and
Enough
the levels
room so no
were indistortion
between
-6dBFS and
-12dBFS,
therefore
leaving
enough
headroom.
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22 October 2015

Looking at the mixes sine wave it is clear that no distortion took place.
As for my third mix, Africa, looking at the sine wave, it is obvious
that so much distortion is taking place. Avoiding this is key to a good
mix.

Compression
Threshold
Compression is used to make loud sounds appear softer and can
reduce an audio files dynamic range.
Threshold can be called the ceiling as when the signal hits the
threshold ceiling, this is the set point where automatic volume
reduction starts. This means that everything above that signal becomes
quieter. A threshold of -10 is much lower and will not have as much of
an effect ton the audio signal as one of -40. Below the threshold, the
compressor has no affect as the compressor only acts on signal on the
other side of the threshold. The compressor reduces the volume to
stop the input signal from getting louder and potentially avoiding
digital distortion.
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22 October 2015

For mix 2, i set the threshold on the left overhead microphone to

-8.0dB and the right overhead to -12.0db. I did this so that the higher
signals coming through from the cymbals and hi-hats, were not
affected or lost within the mix. If i set the threshold on the cymbals and
hi-hats to a high threshold, they would have been muffled and/or lost
within the mix.

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Output / Makeup Gain

A compressor will only adjust the volume when the signal
becomes too loud or gets too high. The Make-up gain comes into
effect and this lets you bring the signal back up to a healthy level and
gives back the dB that was lost from compression. Adding make up
gain will increase the dynamic range of your audio file. For example,
once compression's taken affect, there will be a noticeable different in
volume. If we say that -4dB has been lost, we can turn the whole signal
up 4dB to make up for that loss, This will not affect the compression
already added.

In mix 2, for the electric guitar, I added 1dB to compensate for

the loss through compression. The signal received was already quite
high and so i didn't need to increase the make up gain by that much to
bring it to a healthy level.

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22 October 2015

Ratio
The ratio indicates how much the audio is compressed or turned
down by. For example if the ratio is 2:1, anything above the threshold
is turned down. This means that if there is 2dB above the threshold,
then only 1dB is allowed through. An alternative way of looking at it is
that it has been turned down my 1dB, or 1dB gain reduction. 4dB
above threshold then 2 dB are allowed through, 8dB above, only 4dB
allowed and if 10dB the 5dB are allowed through. Essentially 2:1 ratio
means half.
If we had a ratio of 3:1, every 3dB above the threshold we go,
only 1dB is allowed through and so we will have 2dB gain reduction. If
the ratio was 10:1, then the gain reduction would be 9dB, meaning
that only 1dB is allowed through. If the compressor has a ratio of 1:1
then this means the compressor is doing nothing (neutral).
I set the ratio at 2:1:1 for the kick drum in mix 1. It took me a
number of times to set, what i thought to be, a good ratio as i didn't
want a large amount of volume to be lost.

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Attack and release

The attack time is about how quickly the volume is reduced or
lowered when the input or signal exceeds, or goes over the threshold.
If the attack time is too slow, then for a short time after this, you could
hear bursts of loud, uncompressed, music coming through which
would not be good for your mix.
The release time on compression is about how quickly the
volume can return to normal when the input or audio is no longer
above the threshold. If you allow the volume to come back too fast,
then it will be obvious that the signal level is going up and down just
as if you were turing the volume up and down on a CD player when
listening to music or on a remote when watching television.
The use of a slower attack time, for mix 1, on the snare meant that you
could hear the unique timbre of the snare and help add to the
atmosphere of the piece as the accommodating video is set in a war
zone, hence why the attack time is low at 10.0ms.

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For the release time, referring to the accompanying video, it was best
to have a higher release time, of 68.0ms, to help add to the
atmosphere of the piece as having a slow attack time and a low release
time would cancel out the attack time resulting in a snare sound that is
undesired and does not work for the piece.

Challenges within the mixing process

Challenges i faced within the mixing process included how to
use compression correctly. I found it very hard to judge the threshold,
ratio, attack and release as with compression, it is not only designed to
make loud signal appear softer, but to also, if used correctly, to add
energy into the instrument/mix.
I found compression to be a long task with numerous answers
and faults. For example, finding out what to set the threshold to for
each instrument, then the right/appropriate ratio, attack and release
for it, i found myself going around in circles trying different ratios with
thresholds until becoming satisfied with the outcome. I went searching
for some kind of answer to what do i set the threshold to but all came
back with the same answer which was, there is no definitive answer
but you have to trust your ears and listen.
The Cymbals and hi-hats in mix 2 i still feel are slightly too
overpowering despite the use of EQ and compression on the kick
drum, snare, left and right overhead microphones. I think that i should
have spent more time trying to make them fit within the mix by
experimenting longer with EQ and compression with the different
instruments and microphones.

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22 October 2015

I found it hard to get the electric guitars to sit right within my mix
as i felt that despite playing around with automation, compression and
EQ, they were still very dominant within the mix although their
recording levels were good. I think it did not help that other
instruments were clipping and the guitars had analogue distortion as i
think this made them seem too loud.

How could you improve the mixes/ if i was to do this again.

If i was to do this again, i would have recorded more versions of
the football chant for mix 1 as comparing the audio to the video, its
obvious that the two don't work together effectively and to capture the
scene effectively as one example of how it really didn't work was that
in a football stadium when a game is playing, when people chant, they
are not in time with each other and the audio used was too clean and
in time to use effectively. More versions would have helped along with
a lot of reverb added to the different vocal channels.
For my third mix, Africa, to avoid distortion occurring if i was to
do this again, i would start off mixing the song with the use of EQ and
compressor where needed and once i had done a fair amount to the
mix, bounce the song and take a look at the sine wave to see if
distortion is occurring. If distortion is occurring then i would go back
and try again but if no distortion occurs then i would continue and
repeat the bouncing process after i had progressed further into the
mix.
For mixes 1 and 2, also three if i had been present for the
recording process, i would have asked that each instrument was
recorded separately, or that all DI (direct input) instruments are played
in the same room, and checked their recording levels and used the
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gain on the pre amps, apollo and focus rite to set the right levels for
recording by leaving enough headroom to avoid distortion and
therefore making it easier to mix the songs as i would now have space
to work with.

Bibliography

Image 1.1
From - http://www.planetanalog.com/
document.asp?doc_id=527307
Acesses: 20th October 2015/ Copyright
2013 UBM Tech, All rights reserved
___________________________________________________________
[1] http://www.soundonsound.com/sos/feb10/articles/
analoguewarmth.htm - Accessed: 18th October 2015/ All contents
copyright SOS Publications Group and/or its licensors, 1985-2015.
___________________________________________________________
All screenshots were taken from my work by me and all
information is my own unless stated.

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