How To Get The Best
Vocal Sound Using
ScreenFlow
© 2020 Mike McMillan
INTRODUCTION
Welcome!
Mike McMilan here. This free booklet is a supplement to my YouTube video titled'
"Improve Your Voice On YouTube".
In my video I covered using the following ScreenFlow filters to tweak up the audio on a
sample video I created.
a. Apple: AUGraphic EQ
b. Apple: AUDynamicsProcessor
c. Apple: AUHighpass
d. Apple AULowPass
You may ask, while using these filters, what the "AU" in these filters stands for. AU
stands for "Audio Units", an architecture built into Apple's macOS and iOS operating
systems. The AU platform includes built-in plug-ins such as the ones I mentioned
above.
NOTE: ScreenFlow runs only on Intel-based Macs. You can check out ScreenFlow if
you wish at... https://www.telestream.net/screenflow/overview.htm
ScreenFlow runs $129 to buy but you can download a free trial version to use as long
as you like but all of your projects will then have a Telestream watermark on them.
Additionally, but no less important, I show you how to copy and paste audio filters from
one audio clip to another. This can be a huge, huge time saver for you!
YOUR MICROPHONE
Also, as I mentioned in the video, the higher the quality of your microphone, the better
the quality of your sound. Having said that, I realize most beginners on YouTube aren't
really interested in laying out $500 or more for a very high quality mic. I completely
understand that. And–spending that much on a microphone is not necessary, especially
for beginners.
And that is really the purpose of my video and this booklet: to help give you techniques
to improve your audio quality without spending a fortune on hardware!
Personally, I use an inexpensive lavalier-type condenser microphone which can plug in
to my Canon cameras, my H4N Pro Recorder, or directly into my iPhone.
As I mentioned in my video, lavalier mics can sometimes pick up a muffled, muddy
timber to your recordings because they are fixed close to one's chest and throat. It may
not be noticeable with your equipment but it can be happen. However, you can remove
some of that interference with the ScreenFlow filters I discuss.
Let's look at the filters and techniques I discussed in my video
Accessing The ScreenFlow Filters
All of the filters I used in the video are accessed in the same way...
1. Click on the audio icon in the upper right corner of ScreenFlow
2. Click on the video file you imported or dragged in to the ScreenFlow timeline to
highlight it.
3. In the panel on the right column of your ScreenFlow document, scroll down to
the "Audio Filters" selection and click on the + sign with the circle around it. This
will open the panel just below it with all of the filters shown.
Next click once on any one of the
filters you wish to access to
highlight it. Then click the "Add"
button. (Or just double click on the
filter and it will be added
automatically.)
The panel for the filter you have
added will now appear in the
window in the right. You may need
to scroll down to see it.
That's all there is to it!
You can add as many filters as you wish, however I would add them one at a time. Work
with a filter until you get the effect you want and then add and work with another filter if
you wish. Don't overdo it. One, two or three filters should be plenty.
Filters are processed by ScreenFlow in the order they appear in your audio panel.
Adding a second filter or a third can change the desired effect you obtained with the first
filter slightly.
Fewer filters is generally better. Keeping it simple is good!
Apple: AUGraphicEQ
The Graphic Equalizer allows you to alter the volume of
any frequency within your audio file. The "Hz" you see in
each frequency stands for "Hertz" named after Heinrich
Hertz, a German physicist in the 1800's. A Hz represents
one cycle per second for any type of wave. Thus, a
sound wave with a frequency of 40Hz is vibrating forty
times each second.
20Hz is generally regarded as the lowest (deepest)
frequency the human ear can pick up. At the the other end
of the scale, 20,000 Hz or 20kHz, is generally the highest
frequency our ears can pick up.
While you can use the equalizer as shown at the right to
increase or decrease the volumes of various frequencies
in your audio, I like to use the graphic form shown below.
It's a bit easier to work with, I feel.
Just click on the "Show" button I circled in red at the
top right to make it appear as a pop-up.
You can change the decibel level of any frequency by
dragging any one of the circles up or down. Dragging it up
will increase the volume level of that frequency. Dragging
down will decrease the volume level of the frequency.
Dragging just one circle up or down won't have much
effect that you can hear. You'll likely have to drag 3-4
adjacent circles up or down to hear a difference.
Here is something you may have noticed me do in the
video. If you want to drag any number of adjacent circles
up or down, you can move them all at once with this trick.
A
• •
B
Suppose you want to drag all of the circles between Point
A and Point B all up or down. Don't click on any of the
circles, but rather, click anywhere on the vertical line that
runs through point A.
Hold the shift key down and click on the vertical line that
runs through Point B. A blue rectangle will form covering
all of the circles between those points. Now just click and
hold down on any one of the circles within the rectangle
and all of the circles within the rectangle will move as you
drag up or down.
MAKING ADJUSTMENTS
If your voice sounds a bit to "tinny" then you may want to
drag down a few higher frequencies (at the right side of
the graph). If there is too much bass you can drag down a
few of the lower frequencies (at the left side of the graph).
You can experiment with other frequencies in the middle of
the graph to try different effects.
FLATTEN EQ
In the lower left of the graph there is a "Flatten EQ" button.
If you ever want to reset all of the circles back to 0 dB line
you started on that button will do that.
JUST FOR FUN
Try this. Move your cursor all the way anywhere to the left
side of the graph. Push down on your CONTROL key
and click down on your mouse pad. Your cursor changes to a cross. Keep holding down
on your mouse pad and drag the cross across the graph to the right moving it up and
down as you drag across. Pretty cool, eh?
You can use this trick
to make a large scale
frequency change in
just a second. Play
around with it.
This makes it easy to
test different frequency
settings quickly. Again,
click the Flatten EQ button to get the settings back to the 0dB line.
31 vs 10 BAND EQUALIZER
At the bottom of the graph, next to the Flatten EQ button, you
will see a button saying 31 Bands with up and down arrows
next to it.
In the video I worked with the 31 band equalizer. However, if you click on the down
arrow you can change the graph to a ten band equalizer. It works exactly the same
but you will see only ten circles to adjust. Sometimes it may be easier to work with that
way.
One final thought on the equalizer interface. Perhaps you noticed that the Hz scale does
not increase by an equal amount in each square on the grid as you go from left to right.
Ath the low end it begins at 20Hz, then to 25Hz, then to 31.5 Hz... and the differences
get bigger as you get over to 20kHz (20,000 Hz). The reason (for you mathematical
enthusiasts) is that the Hz scale is what is called a logarithmic scale.
Logarithmic scales are often used when there very large ranges in values on a scale.
For example, the range from 20 to 20,000 Hz in the range of human hearing.
Other common examples of log scales include the pH scale in chemistry, the Richter
scale for measuring earthquake magnitudes and the Decibel scale to measure sound
intensity.
Don't worry, you won't be tested on this (ha ha), I just thought you might have wondered
about that.
Apple: AUDynamicsProcessor
In my video I used the Dynamics Processor to compress my audio file. There is also the
Apple: AUMultibandCompressor in the menu choices, but if you are new to using audio
filters that one is a tiny bit more confusing to work with.
Original Audio File
Compressed Audio File
Above are two waveform files I copied from a ScreenFlow project. The top one is the
original file. Below is the file after I compressed it with the AU: DynamicsProcessor.
While the peaks and valleys in both waveforms appear at the same place, the high
peaks from the top file have been compressed or squished down closer to the lower
peaks.
It may be, depending on your project, that the top waveform is exactly what you want.
But if you want to obtain more uniform levels for the various parts compression will do
the trick.
One key is to not overdo the compression. Too much compression can flatten out
your audio and you will lose the dynamic range which you want to maintain at a
good level.
For example, in music recordings drum or cymbal beats can overpower vocals if
settings were not properly dialed in during the recording. Compression can help tune
down the high peaks produced by the percussion if needed.
At the right I've shown the
interface that appears when you
load the AU: DynamicsProcessor.
I like to use this interface rather
than the graphic form you would
get if you clicked on the "Show"
button. Let's look at what each of
the options does.
COMPRESSION THRESHOLD
This is the decibel level at which you can the
compressor to begin compression.
EXAMPLE: Suppose you highlight your
waveform and set the threshold at -15.
Areas of your waveform with volumes less
than 15 dB will be unaffected by the
compressor. But areas of your waveform with
volumes above 15 decibels will be
compressed.
HEADROOM
This represents the distance between the highest peaks in your volumes and the 0dB
level. Any peaks in volume that hit above the 0bDb mark will be clipped. Clipping will
result in pops and crackles in your audio. YouTube will hate it if your audio has clipping
in your file and if you are having a sound engineer edit your audio in post-production
they will have a mental spasm if you have areas that are clipped in your file.
For YouTube videos having a headroom of around 6dB is good. This will prevent
clipping from occurring and give you some free space above your levels if you have to
apply various filters, which can affect your volume levels.
EXPANSION THRESHOLD
We've talked about using a compressor to bring the high peaks in a wave file down
closer to the low peaks. But a dynamics processor can also do just the reverse. It can
bring the low peaks down even more while leaving the high peaks as they are.
The result is to increase the dynamic range of the audio file, although it can reduce the
low peaks to the point where they can no longer be heard if applied too much. This
would more likely be used in an audio file with musical instruments rather than a spoken
word file.
EXPANSION THRESHOLD SETTINGS
The "Expansion Threshold" setting determines the decibel level at
which expansion will occur. The "Expansion Ratio" determines how
much expansion will occur.
EXAMPLE: Suppose you have a waveform you wish to apply this
filter to. Further, suppose you want to reduce the volume of any
frequencies with levels below 30dB. Click on your waveform to highlight
it. You would set the expansion threshold at 30dB. But then you must
determine the amount of expansion you wish to apply. This is where the
"expansion ratio" comes into play.
Move the "expansion ratio" slider all the way to the left. It will then have a value of "1"
which means no expansion will occur. Then move the expansion slider to the right very
slowly, increasing the ratio only one unit at a time. Notice how the low peaks are
lowered rather quickly as the slider moves to the right. At some point the low peaks will
likely disappear but the peaks above 30 dB remain unchanged.
original wave file
expansion threshold: 30 dB
expansion ratio: 1
expansion threshold: 30 dB
expansion ratio: 2
expansion threshold: 30 dB
expansion ratio: 3
expansion threshold: 30 dB
expansion ratio: 4
You may wonder why they call this expansion when the high peaks don't expand-the lower ones just get lower. Well, that's because the distance between the tops of
the high peaks and the tops of the low peaks does expand. Just so you know.
As I mentioned, I never use the expander function. If you're doing spoken word
videos you probably won't either.
I do, however, see how this could be useful in videos where musical instruments are
involved. Like I said in the video, if you create a video where someone is playing drums
or cymbals, you could have a lot of of transient peaks you would like to bring down. The
expander could do this for you.
ATTACK AND RELEASE TIMES
Any compressor will have attack and release time settings. You shouldn't need to play
with these settings very often, maybe never.
The attack time (these settings are both given in milliseconds--a millisecond is 1/1,000 of
a second) determines how fast the compressor kicks in once the threshold is reached.
The release time is the amount of time before the compressor shuts down once the
threshold has passed. Again, these settings won't likely need to be changed if you are
doing spoken word videos. If you have musical instruments someone is playing in your
video you may find these settings useful.
Apple: AUHighpass Filter
The Highpass Filter in ScreenFlow
does what it's name implies: it
allows high frequencies to pass
through unaffected while
attenuating (decreasing or
eliminating) sounds below a cutoff
frequency you determine.
When you add the high-pass filter
you will see the top interface at the
left. That will work but you get a
better picture of what's happening if
you click on the "Show" button to
give you the graphical representation
shown in the second image.
The high-pass filter is used to decrease low frequency tones and is probably used more
in music recordings than vocal recordings. However you can experiment with it
whichever type of recording you are doing.
By sliding the white dot you can change the blue area under the 0 dB line. The blue
area shows the volume and frequencies that will pass through the filter.
Easy does it with this filter.
In the above graph the cutoff frequency is set very, very high (6900 Hz). A setting like
this will result in a very high-pitched tone with a tinny sound–no good for vocals. You
could try settings in the lower end, maybe up to the 48 Hz mark. A cutoff higher than
that will likely give you a bad sounding wave file.
Of course using this filter will decrease the overall volume of your audio file because
some frequencies have been cut out. You can adjust for that.
RESONANCE
There is also a resonance setting. This setting determines the slope of the line on your
graph at your cutoff frequency. It determines the sharpness of the slope of the sound
around the cutoff frequency. Altering the resonance affects the "cutting slope" of the
filter just after the cutoff frequency (how fast or smooth the high-pass filter engages).
Apple: AULowpass Filter
This filter does just the opposite as the high-pass filter. I'm sure you can figure it out.
From Mike McMillan
I just wanted to thank you for watching my video and for
requesting this booklet. I hope you got some ideas you
can put into play with your own videos.
Remember, if you ever have any questions at all please
don't hesitate posting them in the comment section under
any one of my videos.
And–as always I wish you all the success in the world with
your own videos!
– Mike