DFreqTIP - Finding Glitches With Sound Forge

Sometimes recordings go well and sometimes they don't. During a live

recording, you can't go back and ask the performers to do it again.
So if your live recording suffers any audio problems, you're stuck
with fixing them after the fact. With Sound Forge, this can be a
fairly easy task because it provides a number of tools for fixing
audio glitches. The Find tool allows you locate quick 'pops' and
'clicks' that may occur in the audio. To use this tool, do the
following:

1) Choose Tools > Find.

2) For the Find parameter, choose Glitch.
3) For the Sensitivity parameter, enter 0.
4) Set the Threshold to -60 (this may need adjusting depending on
the level of the glitches).
5) Click OK to find the first glitch.
6) Press CTRL+Y on your computer keyboard to find successive
glitches without having to open the Find dialog box each time.

After you've found a glitch, you can repair it using the various
repair tools that Sound Forge provides. For more information about
dealing with audio noise, read 'Nix the Noise from Your Recordings'
in issue 12 of DigiFreq:
http://www.digifreq.com/digifreq/issues.asp

DFreqTIP - Avoiding Clicks with Crossfading

Someone recently asked me why every time they tried splicing two
pieces of audio together they kept getting clicks in their audio.
They also wanted to know how to eliminate the problem. Well, the
reason this happens is because most of the time the two different
audio waveforms don't line up with one another at the splice point.
Waveforms should be spliced at a point where they both line up with
the zero-axis (a point at which no sound is occurring).

If you have one of my books, you can find a more detailed

explanation on the following pages: Cakewalk Power (pages 154-155),
Sound Forge Power (pages 135-136), Sonar Power (pages 142-143),
Sonar 2 Power (pages 155-156), Sound Forge 6 Power (pages 128-129),
Sonar 3 Power (pages 128-129).

One way to make sure your audio waveforms line up at any spice point
is to use crossfading. This procedure smoothly fades one audio
waveform into another, thus eliminating any potential clicks or
pops. You should be able to find a crossfading function in most good
audio editing applications, but here are the steps to take when
using Sound Forge:

1) Open audio file A and audio file B.

2) Select all the data in audio file B.
3) In audio file A, place the Current Position Cursor at the splice
point (the point at which you want to insert audio file B).
4) Choose Edit > Paste Special > Crossfade to open the Crossfade
dialog box.
5) Choose the Normal Crossfade preset to have audio file A fade out
slowly and audio file B fade in slowly. Or choose the Fast In preset
for a quicker fade. You'll usually want to use a quicker fade if you
don't want the listener to hear the transition between the two
files.
6) Click OK.

[Note: If you are piecing two different songs together, you can do
the crossfade at a point in audio file A (usually the end) where
there is silence.]

Following this procedure, your two audio files should now be one
file with a smooth transition between them. No clicks, no pops, or
other annoying noises.

--------------------------------------------------------------------Q) My audio file has some

background hum. Can I remove it with Sound
Forge?
A) Yes, you should be able to remove it. This is more than likely
caused by some electrical interference or a ground loop in your
audio connections, so the frequency of the hum will be 60 Hz. The
first thing to try is one of the presets in the Paragraphic EQ
function...

1. Select your audio data.

2. Choose Process>EQ>Paragraphic.
3. Choose the 60 Hz Hum Notch Using Four Stacked Filters preset.
4. Click Preview to audition the function and see if the hum is
gone.
5. If the hum is gone, click OK. Otherwise, keep the first filter
frequency set to 60, but try setting the other three filter
frequencies to 120, 180, and 240. The reason for this is that even
though the hum is strongest at it's fundamental frequency (60 Hz),
it can often be accompanied by additional harmonics. Reducing those
harmonics can help remove the hum even more.
6. Click OK.

Q) I have a video file that has an audio track. I'd like to remove
the video data and keep the audio data. Can I do this in Sound
Forge?
A) Sure, just open the video file in Sound Forge, then choose
File>Save As and set the Save As Type parameter to the type of audio
file you need (such as a WAV file). Click Save, and Sound Forge will
create a new file containing only the audio data from the video
file.

Q) Sometimes when I make a vocal recording I get "pops" when the

people say words like popcorn and "ssss" sounds when they say words
like sassy. How can I get rid of these with Sound Forge?
A) The "pops" are called plosives and they occur because of the
quick rush of air hitting the microphone. The "ssss" sounds are
called sibilants. A preventative measure to remove "pops" from your
audio is to use a windscreen while recording, but that doesn't help
with sibilants. Not to worry though, you can use the Multi-Band
Dynamics function in Sound Forge to get rid of both these
problems...

1. Select your audio data.

2. Choose Effect>Dynamics>Multi-Band.
3. Choose the Reduce Plosives And Sibilants preset.
4. Click Preview to audition the function and see if the problems
are gone. If not, you may need to make some adjustments to the
Threshold parameters.
5. Click OK when you no longer hear any plosives or sibilants.

Q) I'd like to make a timed recording in Sound Forge. Meaning, I

would like to start a recording, have it go for a specified amount
of time, and then have Sound Forge automatically stop. Can this be
done?
A) Sure, just do the following...

1. Choose File>New.
2. Set the parameters for your new file, and click OK.
3. Choose Special>Transport>Record.
4. In the Record dialog box, choose Punch-In (Record A Specific
Length) for the Mode parameter.
5. For the Length parameter, type in the amount of time you'd like
Sound Forge to record. You can use the Input Format drop-down list
to specify the time format you'd like to use. For example, you can
even specify a time in Measures & Beats if you'd like.
6. Click the Record button (the button with the big red dot on it)
to start the recording.

Now you can leave Sound Forge unattended and it will create a
recording according to the length you specified, and then stop
automatically.

• Advanced Audio Corrector - "If you record analog audio (such as

recording to a tape), an inevitable consequence is the appearance of
phase shift between the stereo-channels. In the uncompressed
soundtrack these distortions are imperceptible, but if the file is
compressed (such as to MP3), these distortions can be heard as
unpleasant high-frequency sounds. Advanced Audio Corrector allows
you to remove these distortions for the subsequent coding to a
compressed format."
http://www.avlandesign.f2s.com/

Q) Sometimes when I make a vocal recording I get "pops" when the

people say words like popcorn and "ssss" sounds when they say words
like sassy. How can I get rid of these with Sound Forge?
A) The "pops" are called plosives and they occur because of the
quick rush of air hitting the microphone. The "ssss" sounds are
called sibilants. A preventative measure to remove "pops" from your
audio is to use a windscreen while recording, but that doesn't help
with sibilants. Not to worry though, you can use the Multi-Band
Dynamics function in Sound Forge to get rid of both these
problems...

1. Select your audio data.

2. Choose Effect>Dynamics>Multi-Band.
3. Choose the Reduce Plosives And Sibilants preset.
4. Click Preview to audition the function and see if the problems
are gone. If not, you may need to make some adjustments to the
Threshold parameters.
5. Click OK when you no longer hear any plosives or sibilants.

Q) I'd like to make a timed recording in Sound Forge. Meaning, I

would like to start a recording, have it go for a specified amount
of time, and then have Sound Forge automatically stop. Can this be
done?
A) Sure, just do the following...

1. Choose File>New.
2. Set the parameters for your new file, and click OK.
3. Choose Special>Transport>Record.
4. In the Record dialog box, choose Punch-In (Record A Specific
Length) for the Mode parameter.
5. For the Length parameter, type in the amount of time you'd like
Sound Forge to record. You can use the Input Format drop-down list
to specify the time format you'd like to use. For example, you can
even specify a time in Measures & Beats if you'd like.
6. Click the Record button (the button with the big red dot on it)
to start the recording.

Now you can leave Sound Forge unattended and it will create a
recording according to the length you specified, and then stop
automatically.

10. DFreqTOPIC - Massage Your MP3s for Better Fidelity

Before you encode your audio using the MP3 format for uploading on
the Net, you might want to try some of the following processing
procedures to get a better quality sound. These procedures may or
may not produce better results. It depends on the source material,
so always keep a back up copy of your original audio file. But since
encoding to MP3 can reduce the quality of your audio because of the
compression it uses, it may be worth a shot to apply some or all of
the following:

[Please Note: I am using Sound Forge 6, but these procedures should

also work in other audio editing applications. You will need to
figure out the exact steps on your own.]

REMOVE DC OFFSET
1. Choose Edit > Select All to select all the data in your file.
2. Choose Process > DC Offset.
3. Activate the Automatically Detect And Remove option, and click
OK.

APPLY SOME EQ
We want to cut frequencies below 60Hz and above 10kHz. Then we want
to boost frequencies around 200Hz and around 2.5kHz.
1. Choose Edit > Select All.
2. Choose Process > EQ > Paragraphic.
3. Activate Enable Low-Shelf, and set its frequency to 60Hz, and its
gain to -Inf.
4. Activate Enable High-Shelf, and set its frequency to 10,000Hz
(same as 10kHz), and its gain to -Inf.
5. Set up one of the parametric bands with a frequency of 200Hz, and
a gain of +3dB.
6. Set up another parametric band with a frequency of 2,500Hz, and a
gain of +3dB.
7. Click OK.

APPLY SOME COMPRESSION

We want to apply some compression to control the final dynamic range
of the file using a 2:1 compression ratio that starts at -18dB.
1. Choose Edit > Select All.
2. Choose Effects > Dynamics > Graphic.
3. Set the Output Gain to 0dB.
4. Activate the Auto Gain Compensate option.
5. Set the Attack to 1ms.
6. Set the Release to 500ms.
7. Set the Threshold to -18dB.
8. Set the Ratio to 2.0:1.
9. Click OK.

NORMALIZE THE AUDIO

Finally, we want to normalize the file to get the highest amplitude
without clipping in order to mask any possible encoding artifacts.
1. Choose Edit > Select All.
2. Choose Process > Normalize.
3. Activate the Peak Level option.
4. Set the Normalize Using parameter to -1dB. Don't set it to 0dB
because the encoding process may need that small amount of dynamic
range to render the file correctly.
5. Click OK.

Now you're ready to encode your file to the MP3 format. As I

mentioned earlier, these processes may or may not help in getting
better fidelity for your audio files because the source material is
always different. If you want to make a comparison, first encode
your original file as an MP3. Then take the original file and apply
the aforementioned processes, and encode that file to MP3. Now play
both files and listen for whichever one sounds better.

--------------------------------------------------------------------
********************************************************************

10. DFreqTIP - Eliminating the Lead Vocal

No, I'm not talking about bumping off the lead singer in your band.
What I mean by the title 'Eliminating the Lead Vocal' is removing
the main vocal part from a prerecorded song. Someone recently asked
me if this was possible, and I told him yes and no.

Yes you can remove the lead vocal from a prerecorded song, but only
if the vocal is panned directly in the center of the stereo field.
And even then, the process isn't perfect. There isn't currently any
audio software on the market that can analyze and remove only a
single vocal part. Instead, you have to cut out the material in the
middle of the stereo field. That means cutting out the vocal and
everything else centered in the field.

Here's how to do it in Sound Forge 4.5 and 5.0:

1) Open the audio file.
2) Choose Edit>Select All to select all the data in the file.
3) Choose Process>Channel Converter.
4) In the Channel Converter dialog box, choose the Stereo to Stereo
- Vocal Cut preset.
5) If you want to end up with a pseudo-stereo file, no other
settings are necessary. If you want to end up with a pseudo-mono
file, activate the Invert Right Channel Mix option. Using this
option makes the final results sound a bit better.
6) Click OK.

Here's how to do it in Cool Edit Pro:

1) Open the audio file.
2) Choose Edit>Select Entire Wave to select all the data in the
file.
3) Choose Transform>Amplitude>Channel Mixer.
4) In the Channel Mixer dialog box, choose the Vocal Cut preset.
5) If you want to end up with a pseudo-stereo file, deactivate the
New Right Channel Invert option. If you want to end up with a
pseudo-mono file, no other settings are necessary. As I mentioned
earlier, the mono file sounds better.
6) Click OK.

Depending on the type of material that you're processing, the

results will vary, but they'll never be perfect. You can try to
tweak the mix a bit with EQ, but other than that, what you get is
what you get. Maybe in the future we'll have more sophisticated
software available that will be able to analyze and separate
specific sounds from an audio mix.
Five Sweet Sound Forge Jobs
By Jeffrey P. Fisher
Here are five audio tasks ideally suited to Sound Forge.

Clean up DV audio

If you have video shot with the on-camera mic, it's going to need some
massaging to sound good. Import the audio track into SF and get to work. I
usually start with the Process> EQ > Paragraphic and check both the Enable low
shelf at 100 Hz and -inf and high shelf at 10,000 and -inf. This gets rid of
unnecessary low and high end noise. If there's a troublesome frequency band,
say from some machinery, I'll find it with this EQ by boosting, then cutting once I
hear the offending sound. Often I'll put a little 2-4 dB bump between 2.5 and 3.5
kHz to make speech more intelligible. Next, I'll even out the levels with some
dynamics processing. Start with the Effects > Wave Hammer preset, "Master for
16-bit" and tweak from there. You want your file louder and even. If you have it,
use the Noise Reduction plug-in to cut back on extraneous noise. You might be
able to follow up that treatment with Effects > Noise Gate and smoothly eliminate
the background (except when the voice is speaking).

Record narration

If you just need a wild voice-over (not synced to video action), put SF to work. I
use the "Multiple takes creating regions" record mode which automatically
creates regions for each take. That provides a visual guide when editing the file.
You can also throw a marker while recording to designate a specific section.
Once recorded, I'll clean up the takes in Sound Forge before taking them to my
multitrack/video NLE (Vegas!)

Play list
Here's an extension of idea 2. Click View > Playlist to display the Playlist. Now
double-click any region to select it and then drag it to the playlist. (If you have no
regions, select sections you want, and press R). Once you've chosen the pieces
you want, right click the playlist and choose Convert to New to instantly create a
new file using only those regions/clips you selected. This method is by far the
fastest way to edit a VO track (deleting mistakes, choosing the best takes, etc.)

Sound design

I love to create new sounds (and mangle existing ones). My favorite tool for this
is Acoustic Mirror. You can superimpose the characteristics of one sound on
another. First, select the sound you want to mangle. Next, choose Effects >
Acoustic Mirror. Select the Impulse Browse function and hunt down a .wav file to
use. In preview mode, tweak until you get something you like. There are a bunch
of impulses on the Sony Web site, but these are more reverb-oriented. I'm
suggesting using Acoustic Mirror with other sounds. Try it!

Mastering audio for CD/MP3

When I've finished mixing a music piece, I drop it into SF and make it shine. First,
clean up heads and tails to prep the song for mastering. Then, I'll use EQ,
multiband compression, and the Wave Hammer to add that extra sheen to the
track. Izotope Ozone is another mastering tool that works (and sounds) great in
SF.

ALIAS FILTERING

As discussed earlier, raw analogue audio signals can contain a very wide range
of frequencies -- they don't stop at 20kHz just because we humans can't hear
above that (if, indeed, many of us can hear as far up as 20kHz anyway!). We
have also seen that we need to restrict the frequency range in a sampled audio
system to comply with the Nyquist theorem. This implies that some severe audio
filtering is required to restrict the upper frequencies being input to the sampling
process, as well as to remove the unwanted image frequencies from the output
signal.
If our sampling system operated at a 40kHz sampling rate and the original audio
input happened to contain a signal at 30kHz (which would be inaudible to
humans, but could still be present), the lower image of that signal would appear
at 10kHz (See Figure 5 opposite). Not only would this be clearly audible, it would
also be impossible to separate the unwanted image frequency from the wanted
audio band.
This effect, where an unwanted signal appears in the
wrong place, is called aliasing -- and it sounds most
discordant. The best description I can muster is that it
sounds a little bit like the 'birdies' which can beset poorly
aligned FM radio receivers sometimes -- a kind of
unrelated and unmusical chirping which, in extreme circumstances, can make the
finest piano sound like a very rough harpsichord!
In analogue systems, distortion produces additional signals, above the original
audio components, which are related harmonically -- as in second or third
harmonic distortion. These may be undesirable, but being 'musically' related to
the source they tend to be acceptable -- even beneficial -- if sufficiently mild.
However, aliasing in digital signals produces tones which are not musically or
harmonically related to the source signals. Instead, they are mathematically
related to the sampling rate and the aliases appear below the signal which
caused them. Since this is a very unnatural phenomenon, the ear can detect
aliasing even when there is only the smallest trace of it, and it is particularly
unpleasant.
The visual equivalent of aliasing is strobing -- the classic example being the
wheels on the stage coach in an old western movie which appear to be going
backwards slowly when in reality they are going forwards very quickly! The true
rate of rotation is too fast for a film camera (with its slow 24 frames-per-second
'sampling rate') to capture, but the 'alias' of that rate is visible as a much slower,
often reversed, rotation.
As it sounds so unpleasant, it is essential that aliasing is not allowed to occur in
an audio sampling system, and the original analogue audio is low-pass filtered to
ensure that nothing above half the sampling frequency can enter the system.
This filter is called an anti-
anti-alias filter (since that is what it filters!), and it prevents
the lower image and the audio signal from overlapping.
The sharpest low-pass audio filters normally encountered on mixers or synths
might have rolloff slopes of up to 24dB/octave, which means that for each
doubling of frequency (ie. octave rise in pitch), the signal is attenuated by a
further 24dB. However, to achieve sufficient isolation between the wanted audio
signal and the unwanted lower image in a sampled signal, we need perhaps 90
or 100dB of attenuation for an anti-alias filter -- and there is typically only a
fraction of an octave to do it in! Anti-alias filters therefore have to be extremely
steep, with rolloff slopes in the order of 200 or 300dB/octave, and in order to
allow them sufficient space to achieve a useful degree of attenuation, the Nyquist
theorem requires that the sampling rate is actually around 2.2 times the highest
wanted audio frequency. (See Figure 6, above.)
As mentioned earlier, a very similar filter is required on the output of a digital
audio system to remove ultrasonic images, left over by the sampling process,
from the wanted audio signal. Although inaudible by definition, if left in place the
very high image frequencies would quickly fry the tweeters of loudspeakers,
could potentially cause instability in amplifiers or other electronic circuits, and
would cause interference with the high-frequency bias signals on analogue tape
recorders. The output filter that removes these high frequencies is called a
reconstruction filter and design economies usually mean that it is identical to the
anti-alias filter on the input of a digital system.
How to fix gradual loss of sync once and for all

Ok i guess this topic has been covered quite a few times. So i'll try and make it quick.

Now Phat J's guide did give me some clues, but he overlooked a simpler, more accurate
and quicker solution. I guarantee and promise this will work. Basically you need to
resample rather than compress or expand the time. The example I've done here works for
MPEGs but you could apply the same methods for AVI files.

Unbeknownst to most people I should imagine. Sound cards suggest they are record at
44.1KHz. Well in most cases they probably do. However, sometimes they don't quite
sample correctly and are off by about 0.005KHz. This is what causes your gradual sync
people. Some guitarists may recognise this if you have a software tuner. You need to
calibarate the soundcard to get the perfect tuning.

The method will also work with VHS captures too, even if you capture other sources and
they are in sync.

Is everybody in, is everybody in, so let me begin.

You will need:

Sonic Foundry's Sound Forge or Cool Edit (Both are available as demos)
TMPGenc
VirtualDub

1. Similiar to Phat J's guide

Use VirtualDub to find a time in the video to find a noticeable sound. It helps if you can
lip-read but it's not essential. Make a note whether the sound is behind or before the
video.

Here I've found a frame from the Let It Be film, where John is going to sing the word
"Seeeeee" from the song I've Got A Feeling.
Jot the time down, in this case it's 1:16:22.891. The audio is ahead.

2. Use TMPGenc and under MPEG Tools to demultiplex the MPEG file into two separate
files. So in this case it would Let It Be.m1v for the video and Let It Be.mp2

3. Open your audio file in Sound Forge (This may take a while if it's huge) and find your
jotted time. Play the sound and note the where it's playing.
Here you can see the beginning of the "Seeee" peak and the audio is about 0.939 seconds
ahead out of sync.

4. Now here's the amazing part. Instead of expanding or compressing the sound, we're
going to adjust sample rate of the sound instead. Under Process you will find an option
called Resample.
Make sure you only set the sample rate first only. Here i've added 9Hz because the sound
was ahead so it needs to be played ever slightly more quickly. Reduce the sample rate if
it's behind. You'll have to tinker with the sampling rate, but you'll find that adding 1 Hz
will reduce about 0.1 seconds off over an hour. Don't worry about the pitch change, it's
doesn't make a difference due to such a small change.
Now you can see i'm only 0.002 seconds out. This is alot more accurate than the time
expand/compression utility. Not only that, you don't have to render the effect to preview,
so you don't have to wait ages if you've got a big file. Some soundcards won't play at
41,109Hz but this really isn't a worry.

5. Once you've found the correct sample rate, you need to resample. So this time go back
to Process and Resample and clear the box that says: "Set the sample rate only(Do Not
Resample") and adjust the new sample rate to 44,100Hz. You don't need to apply an anti-
aliasing filter, the resampling difference is so small to justify the time taken to apply the
filter. Once you've set it it will start to resample which may take a while.

Now you can see your sound is at 44,100Hz and the sync is damn as near correct as it can
be. No amount of time expanding/compression can get this close.

6. Now all you need to do is save the file. You'll have to save it as a Wave File at
44,100Hz Stereo unfortunately, which means mega space, but it'll be worth it.

7. After you've saved it, use TMPGenc to change the wav file into an mp2 file.
8. Use TMPGenc again and goto MPEG Tools and select Multiplex. Here it would be Let
It Be.m1v as the video and Let It Be.mp2 as the audio. Choose anything you want as the
output file( well make sure you end it .mpg or mpeg)

9. Enjoy your newly synched video.

10. Praise me for wasting hours trying to sort this out once and for all. I really do believe
you can't get any more accurate than this method and it works with humungous files.

So to cap off, blame the soundcard, it's not recording at the correct sampling rate it's
supposed to. The soundcard in this case actually records at probably about 44,109 to
44,110Hz. Perhaps video capturering software needs to use an audio calibrating utility
like guitar tuning software does to help with the sampling rate and synch.

Thanks to Phat J for spurring me a bout of inspiration and thanks all for those in sync-
hell. Welcome to uniformity.

Yours in humanity

ACID Agent
Audio Mastering

Audio mastering is one of the essential arts you need to learn to make your
tracks sound good on CD, vinyl or even as MP3. People often confuse mastering
with mixing, but the two are different in many ways : Mixing is balancing the
levels between instruments and getting the individual instruments to sound good,
where as mastering is the final step where you want to polish the overall sound
and maximise volume. Often the reason that commercial CDs sound so much
louder than your own mixes is that they use compression and clever limiting
techniques to maximise the levels, boosting the overall sound. However, with the
right tools you can do this in a home studio.
If you're looking for a far more detailed explanation of what mastering is and why
it is needed then read this excellent article at Digido.Com.

Tools You Will Need

To master a track you will ideally need a decent sound editor (preferably one that
will work at high bit and sample rates and will accept DirectX and/or VST
plugins), such as SoundForge, WaveLab or CoolEdit Pro. There are also stand-
alone packages dedicated just to mastering, such as T-Racks. However, if you
haven't got one of the 'big three' then check out HitSquad : Shareware Music
Machine for a list of free and shareware audio editors. Ideally you will also have
some high quality plugins for your audio editor for mastering with (as the ones
that come with audio editors are sometimes lacking) - the best ones I know of are
made by Waves (their Native Gold Bundle has everything you need). At the very
least you will need to be able to normalise your audio, EQ it and compress it.
Ideally you should also have access to plugins that will remove DC offset and
also a multi-band compressor, parametric EQ, stereo imager and a limiter.

The First Steps

The first thing you need to do before mastering audio is make sure you are
happy with your mix! It's very difficult, if not impossible, to fix errors in your mix
when you are mastering, so make sure your mixdown is as good as it can be
before exporting it to an audio file for mastering. Preferably listen to it on good
monitor speakers (after resting your ears) at different volumes to make sure that
everything sounds balanced and that your bass frequencies are prominent but
not too 'boomy'. It's also a good idea to listen to your mix on cheapo PC speakers
too, so you know how it will sound on more basic setups - again make sure that
everything still seems balanced on crappy speakers.
Once you are happy with the mix the next step it to export it as an audio file.
Before doing this, though, make sure your mix isn't clipping - when you are
working in the digital realm you NEVER want your maximum level to exceed 0dB.
Don't worry if your mix sounds a little quiet, that can be solved in the mastering
stage - just don't be tempted to let it clip. OK, now you need to actually export
your track as audio (normally this will be as a .wav file on PC or as an .aiff file on
Mac). Often you will get a choice of what resolution to export your audio at ie. the
sample and bit rate. Generally it's best to choose the highest bit and sample rate
your audio software will support - 24bit/96Khz is usually the best quality, if you
are given the choice. Do this even if your audio card doesn't support playback or
recording at these sample rates (this might sound counter-intuitive, but trust me it
will still work!). It's probably beyond the scope of this short piece to explain why
you should always master at high sample-rates, but basically it's down to
reducing errors caused by floating point precision - the higher the precision the
less chance of errors creeping in. For a full explanation have a read of this article
on mastering and dithering at digido.com.

The Mastering

OK, now you've exported your audio file you will now need to open it in your
favourite sound editor. The first thing to do is remove any DC Offset. Next you
want to EQ your audio file - this will be a matter of personal taste, but now is the
time you can boost the bass or add a little more 'air' to the mix. It's also a good
idea to roll-off any inaudible, low frequency bass sounds - usually a high-pass
filter set to roll of frequencies below 60hz will do. This will help clear up your
bottom end and avoid things sounding muddy, especially on systems with sub
woofers. If you have any plugins such as Waves MaxxBass now would be a good
time to use them. For more info on EQing have a read of this article on Songstuff.
For a guide to what EQ frequencies to tweak to help specific instruments come
through in the mix look no further than another excellent SongStuff article.
Next you should look at compressing your mix - this reduces the peaks and
allows you to increase the overall amplitude, or loudness, of your mix. If possible
use a multi-band compressor which allows you to add different levels of
compression to different frequencies. If you're not sure about compression then
read more about it's uses here and here. Just remember to avoid clipping, as
digital clipping is nasty! If you have access to one a limiter is very useful (a limiter
is basically a 'brick wall' compressor that stops a signal ever going beyond a
defined threshold - typically set this threshold to -0.3 dB for CD mastering). A
great limiter is the Waves L1 or L2 Ultramaximizer. Remember you should put
your limiter last in any audio chain. Also be aware that once you have raised your
overall level to very near 0dB that you should NOT do any more processing on
the audio else you risk introducing clipping (believe it or not, even subtracting EQ
can actually increase levels).
Beside EQ and compression there are other tools you can use too, depending on
the effect you wish to achieve. Sometimes adding a very small amount of reverb,
especially one that defines a real 3 dimensional space, can help bring your mix
together. This will have the effect of situating your track in a virtual sound-scape.
For instance, a touch of room reverb might be useful to bring a rock mix together,
whereas for more ambient, experimental works a larger reverb could help expand
the sound. Just don't over do it, especially if you've already used a lot of reverb
when mixing, otherwise your mix will sound mushy. If you can limit the reverb to
just the higher bands this would be preferable, as you don't want your bass and
kick drums to sound boomy. Another useful tool is a stereo-imaging tool that can
help give your mix a wider stereo field. This is useful for mixes that sound a bit
too mono, or when you are making vast electronic music soundscapes. Again,
easy does it, as too much can make your mix sound strange and the bass
sounds weak.

Stop Dithering!

One final thing you'll have to take into account if your mastering digital music is
dithering and re-sampling. No, I'm not referring to the usual procrastination and
uncertainty you have when mixing, but rather the process of getting your final
master into the right format for burning to CD. As you probably know CD Audio
uses a sample rate of 44.1Khz and is 16 bits deep - so what do you do to make
sure your pristine 24bit/96Khz master sounds good at this lower rate? Well, this
is were dither and re-sampling come in. Basically this is the process of reducing
the resolution of audio by aliasing the waveform - it's rather similar to when you
shrink a photograph in Photoshop and you use bi-cubic resampling to remove the
jaggy edges you'd otherwise get. Good dithering plugings introduce a small
amount of noise to help smooth the audio 'jaggies' when the bit depth is reduced
(this is often referred to as noise shaping). Now, adding noise to your master
doesn't sound like a good thing, but believe me this is virtually inaudible and
really does help maintain the quality when you dither down. It's something you
need to if you're mastering for CD, but make sure you do it at the final when you
are fully happy with your mastered mix. For more technical information on
dithering I'd recommend reading this great article and this one too.

Useful Links

• ComputerMusic mastering tutorial - A beginners guide to mastering audio using

software
• Sound On Sound mastering article - Informative article about mastering audio on
your PC from SOS Magazine
• www.digido.com - One of the most through guides to audio mastering on the web,
but quite technical in parts
• Tweakheadz audio plugins guide - Good article with links to some of the best
plugins for mastering audio on PCs
• www.waves.com - Developers of some of the best audio plugins
• TC Works - Developers of the famous Native Bundle plugins
 • DirectX Files - Comprehensive directory of DX audio plugins, including quite a
few free ones
• HomeRecording.Com - A (fairly basic) website devoted to home recording
• Dither Explanation - Excellent article by Nika Aldrich
• The Cause Forums - Cool, relaxed forum discussing music production

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