The Fundamentals of Sound in Post Production

The Fundamentals of Sound in Post Production


Hi, John Hess from FilmmakerIQ.com and today
we’ll get into the post production side of audio – establishing fundamentals and looking
at digital audio workstation tools for mixing and perfecting your film’s soundtrack. In this lesson we are going to be diving deep
into shaping sound so it’s important that you have a fair grasp of the different aspects
of sound – You may want to review, if you haven’t already, our video on the Science
and Engineering of sound as we will be using many of the terms laid out in that lesson.
For this lesson I will be demonstrating with the tools that are inside the Adobe Creative
Cloud including Premiere Pro and their Audition. All the tools I mention here should be available
in other audio editing programs and many may be included in your NLE of choice. The first tool or weapon in the sound editor’s
arsenal is the equalizer. But what exactly is an equalizer? In real basic terms an equalizer boosts or
cuts the amplitude of certain frequencies which alters the harmonics or overtones resulting
in the change of the character of the sound. Let’s imagine the audio response of a wave
as a straight line on a graph where the x axis represents the frequency going from low
to high and the y axis represents amplitude. Now let’s say we want to boost ONLY the high
frequencies – say everything above 5,000 Hz. Our straight line is now broken into two levels
with a slope in between: this is called a high shelf. This type of equalization, called a first
order filter, is the simplest kind of equalization to perform using electronic components. This
is found on your basic consumer hi fi systems. To continue, let’s say we want to cut the
sound of the low frequencies in our recording below 100Hz – our line reflects that with
a low shelf cut. Now if we go to the extreme and eliminate
all sounds from above or below a certain frequency, this shelf is called a pass filter – a High-pass
filter essentially lets all the high frequencies pass, eliminating all the low range, where
as a low-pass filter does the opposite – let all the low range pass and killing off the
high frequencies. But what if we want to target a more specific
range of frequencies? That’s where second order filters come in. This is often called
the peaking filter or parametric equalizer and it has three settings: The frequency,
which is what frequency you wish to target, the gain: how much you want to boost or cut
that frequency, and the Q or quality factor which is how wide the parabola of the adjustment
will be. High Q values will have a steeper slope. Sometimes Q is expressed in octaves
– the more octaves a Q has the more wider and gentler the effect. A really high Q filter are used to completely
eliminate a particular frequencies is sometimes called a notch cut or a band-stop filter.
These are used to eliminate constant frequency based noise like a electronic hum or to prevent
feedback in a live audio setting. Another type of equalizer you may come across
are Graphic equalizers. These will be commonly found on mix boards, they behave the same
way as parametric equalizers except instead of selecting specific frequencies and changing
the q value, all the frequencies are presented as sliders with a predetermined interval and
q value. So how and why do we use equalizers? There
are essentially three main uses- first is to fix inadequacies in the recording: Microphones
aren’t perfect and some have a specific frequency response and you may want to use the equalizer
to compensate and create a flatter response. You can also target specific hums with a notch
filter and eliminate them or use a high pass filter to cut low range rumble caused by wind
noise. The second use is when you’re mixing audio
sources that are competing in a similar frequency space – a common occurrence when mixing voice
over with a background music track, if you cut the background music in the 1200 HZ range,
the sweet spot of human voice, you can make some more room for dialogue or voice over
tracks: The final reason and arguably most important
use of EQ is for creative reasons making the track sound better – or just different. For instance boosting the bass frequencies
on a dialogue track, say around 160hz will add power to human voices -but too much can
make the track muddy and unintelligible. You can add a bit of presence by boosting the
5kHz range but again too much will cause ear fatigue. The sibelence or �ess� sounds
can be found between 4 and 10 kHz, you can boost this for more of a clear sound or cut
it to get rid of harsh ess sounds. Playing with these different EQ settings will get
you closer to your desired sound. If you’re mixing instruments – there are many charts
available online that give you a general guideline for which frequencies to target depending
on the instrument. You can even go further and push EQ to create
brand new sounds. For instance – EQ can also be used to simulate the sound coming from
a radio or walkie talkie: Houston I think we have an EQ problem. In music, dynamics refer to the general loudness
of a passage from piano which is soft to fortissimo which is loud and forceful. Dynamics in sound
engineering is same concept – the dynamic range is the difference from the very soft
to the very loud. Now sometimes we need to compress that range – to make the difference
between soft and loud passages smaller. This is the work of a tool called the compressor.
To visualize what the compressor does, let’s use, what else, a graph. On the x axis we’ll
put our input level in decibels and on our y axis will be our output level. If we don’t apply any compression at all we
will have a straight line curve going 45 degrees up the the chart with a slope of 1. For any
given input, the output will be exactly thesame. A compressor works by essentially squashing
down sound that goes above a certain threshold – let’s say we we want to dampen everything
that goes above-12dB. A compressor essentially draws a new line starting at -12dB this time
with say a slope of one half – or 2:1 compression. This means for ever 2 dB increase in volume
above -12dB from the input, there will only be a 1 dB increase in output volume. A more
drastic compression would be 4:1, for each 4dB increase of input there would only be
one dB increase in the output. Compressors have settings for attack and release
to determine how quickly or slowly they kick in. Too fast and you can get a pumping sound,
too slow and spikes in the audio can slip through. Once we have compressed the dynamic range,
we can safely boost the entire track to make everything generally louder if desired. If you push the slope flatter to say 20:1
or 100:1 you get what is called a limiter. A limiter essentially prevents peaks from
going over a specific target generally used for broadcast and they have very short attack
and release times. The opposite of a compressor is called, as
you might imagine, an expander. Going back to our curve – an expander is a part of the
curve that has a slope of greater than 1. Let’s say we want the audio that is below
-20 dB to get quieter faster – our curve reflects that with steeper slope. Expanders are generally only used for the
quieter parts of the dynamic range. A noise gate is one kind of expander – a noise gate
is essentially like a high pass filter except for amplitude. Anything louder than the threshold
will get through, anything lower than the threshold will be expanded down into nothing.
Attack and release settings are available for expanders as well and need to be tinkered
with to find the best settings. So why would we need to use a compressor?
Compressors help smooth out sudden increases in volume caused by momentary changes of distance
from the mic or just natural changes in volume. Smaller dynamic ranges may be necessary for
your venue – if you’re mixing audio for a video that will be shown on the floor of a
subway station there’s going to be a lot of ambient noise and you’ll need to boost the
soft parts in order to compete with that noise. Which gets us to one of the main use of compression
– to make the audio sound more powerful and louder than it really is. Over the years the
recording industry has move towards making their albums sound as loud as possible – comparing
the waveforms from a recording from the 70s and a modern song show just how much compression
is used these days – that’s not always a bad thing as people are often listening to music
in their cars or on earbuds where it’s important to keep a consistent level while having that
feeling of loudness. A really handy tool for bringing out more
life in an audio track is the multiband compressor – it essentially combines the best of EQ – the
control of harmonics and overtones with the control over dynamic range that a compressor
has. A multiband compressor essentially breaks the track into different bands of frequencies
which you can independently apply compression. For example on voice tracks you can compress
and boost that 160hz range for adding power while leaving everything else alone. Most
programs will have several presets to pick from and I almost always find myself reaching
for the multiband compressor when finishing my mixes. Expanders can be used as noise gates which
can push our noise floor lower but there is another technique in the digital realm for
eliminating noise which is called the Fast Fourier Transform or FFT. Inside Adobe Audtion, FFT is a stand alone
filter or part of their noise reduction suite and it works by first taking a snapshot of
your audio waveform – creating a profile of the unwanted sound. Then using various settings
you can subtract the offending noise from the entire track. Now the problem with FFT processing is too
much can result in something called chirping which is squirrely weird digital bird sounds.
You can avoid chirping but not completely removing background noise. To my ears a little
bit of ambient noise is not necessarily unwanted as it can give a little warmth to a track. But FFT isn’t used only noise reduction as
you can use it to remove practically any sound from car horns, to footsteps to instrument
hits. There are a lot of neat and amazing things can be done with FFT. Now we get into probably the most fun filters
– certain the first ones I tried out as a kid first playing with a digital audio program. Using a delay filter – we can create some
really interesting effects. By repeating the audio with a delay of 15 milliseconds or less,
we get an effect called combing where interference patterns created resemble that of a comb.
Now combing is generally avoided in the recording stage, it’s caused by quick slappy echo but
as an effect it may be able add something unique and interesting to the mix. With a delay of 15-35 millisounds we start
getting chorusing effects where the brain is starting to perceive more than one voice
or instrument is being sounded. Chorusing filters can also vary the pitch and timing
of the delays for more effects. This may be useful for creating bizarre and other worldly
characters for your audio Beyond 35 miliseconds and we will begin to
perceive an echo effect. Along with echo are reverb filters. Instead
of being a direct delayed copy, reverb is the mixture of a large number of random and
decaying echos. Advanced digital reverb generators can even simulate the time and frequency response
of a specific rooms like concert halls. Echo and reverb can give your audio track a sense
of space – whether that’s a large cavern or even a small hard room. Now if we take a wave and we squeeze the time
we are by very essence adjusting the frequency. Make the time shorter and the frequency will
go up. Stretch it out longer and the frequency will go down. This is the most basic form
of pitch shifting and it’s sometimes linked to the Chipmunk Effect – where the original
songs were sung at half the speed and an octave below and then played back at twice the speed. But say you want to change the time of the
track without chanigng the pitch – or vice versa. To do this, audio programs use either
Phase Vocoders or sinuosodal spectral modeling to stretch and squish waveforms making things
like auto tune possible. These essentially model the new desired sound frequency waves
using rather complicated math which we’ll just leave to the audio engineers and programmers Wow, we’ve barely scratched the surface of
what goes into audio engineering and sound design. But the whole purpose of this lesson
has been to provide some foundational ground work from EQ, Dynamics, Noise Reduction, and
Time and Pitch effects, I hope this has cleared up some of the mystery of working with post
audio. No filmmaker, no sound mixer, or artist working in any medium, can simply watch a
video or take a class and suddenly become the top of the field. It takes practice practice
practice. And in the case of audio mixing, a lot of time just fiddling with those knobs
and buttons and experimenting with how it sounds when you boost this range or cut that
frequency. Experiment, play and repeat – if it sounds good, and you have a decent pair
of speakers – then it IS good. Don’t be afraid to try things and fail, because it’s all on
the path to making something great. I’m John Hess, I’ll see you at FilmmakerIQ.com

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