Clean Up Your Recordings (CM 197)

34 / COMPUTER MUSIC / November 2013

YOUR RECORDINGSYOUR RECORDINGSYOUR RECORDINGS

Whether you’re looking to fix a

dodgy recording, restore an old

demo track or just do away with

irritating mains hum, our

curative walkthroughs will

patch up all your audio injuries

November 2013 / COMPUTER MUSIC / 35

clean up your recordings / make music now <

Picture the scene: after a long day recording in an unfamiliar studio,

you get home for a playback session on your trusty monitors, only to find that things aren’t sounding as great as you thought. The bass is completely lacking in bottom end, the rhythm guitar sounds dreadfully harsh, there’s mains hum all over the keyboard parts and the drums sound like a set of suitcases falling down a flight of stairs. Suddenly the whole thing is a disaster. But don’t panic – ’s here to make it all better!

There are plenty of reasons for recordings “going south”. A lack of experience on the part of the engineer is a common one, but poor performances, phase issues, time restrictions, dodgy equipment, loose connections, broadband noise interference and numerous other gremlins can also ruin the results of an otherwise successful session.

And, of course, these issues aren’t restricted to those recording bands. Perhaps you’re looking to resurrect a noisy old demo from tape, or you’re working on a mix or remix using parts that have been recorded and sent to you over the net by other musicians/producers, with highly variable results. Or you’ve got a sampled drum loop

you want to use in your latest electronic track that grooves like a mutha but is ruined by weird resonances…

Ultimately, what we’re trying to impress upon you is that every computer musician will find themselves needing to rescue a bad audio clip sooner or later – and that’s where we come in. You may not realise it, but your computer gives you access to some extraordinarily powerful and sophisticated tools with which to repair audio (and we’re not just talking about pricey restoration plugins here). While there’s no sure-fire, single-click substitute for getting things recorded properly in the first place, the phrase “we can fix it in the mix” has never rung more true, and there’s an abundance of tools and techniques that can rescue your tracks.

Over the pages that follow, we’re going to diagnose and cure a whole host of ailments that you may commonly encounter in your audio engineering and production endeavours, both purely sonic and performance-related. You’ll find all the examples and files accompanying these walkthroughs on the DVD and at vault.computermusic.co.uk, so scrub up, grab your audio bone saw and prepare to operate!

How to use this tutorialTo demonstrate our techniques, we’ve created a simple demo recording project that we’ll be using for all the walkthroughs. The multitrack audio for this project can be found in our Tutorial Files folder, which you can download over at vault.computermusic.co.uk or find on the DVD that comes with the print edition.

The easiest way to follow along is to make a new project in your DAW, set it to 144BPM, then drag the WAV files from the _STEMS folder onto separate tracks in your DAW. You’re now ready to follow our tutorials!

Get the samples, video and

tutorial files on your PC/Mac at

vault.computermusic.co.uk

DOWNLOAD


> make music now / clean up your recordings

> Step by step 1. Smoothing out harsh guitars with EQ

We’re going to fix a thin, harsh guitar track with too many upper-mid

frequencies, although this technique will work just as well to correct signals that are dull, boomy, etc. We can pinpoint the problems using a spectrum analyser plugin, such as Voxengo SPAN (free at www.voxengo.com). Set your project to 144BPM, import the test guitar from the _STEMS folder, and add SPAN to its track.

1Click the Edit button and set the meter type to Cumulative Average (AVG),

which gives an overall picture of the frequency levels over time, rather than a constantly updated ‘dancing’ graph. For greater accuracy, set a high Block Size, and turn up the Overlap to prevent the display blinking. Then set Smoothing to 1/6 Oct – this makes the display easier to make sense of.

2Play a long section through the analyser – the longer, the better, as this

gives more of an overview of the general guitar tone rather than just showing the harmonics in one phrase or chord. There’s a big peak around the 3kHz region, which gives the guitar that ‘honky’ harshness. There are also troughs around 250Hz, 550Hz and 1.7kHz. Note that the high-end roll-off seen here is normal for guitars.

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The resulting graph is much flatter, and the guitar now sounds much

fuller and more mix-ready. Troublesome resonances at specific frequencies (‘whistling’) can be isolated by disabling Smoothing and zooming in the analyser display to see if anything pokes out. We’re looking for tall, sharp spikes at precise frequencies – here we can see that there’s one at around 1.05kHz.

5Now we insert a parametric EQ before the analyser in the plugin chain, so

that the analyser displays the EQ’d sound. We’re using DMG Audio’s EQuality, but any fairly flexible EQ with a good number of bands would work just as well. Using the EQ, we boost the areas where the analyser shows gaps, and make cuts where there are peaks, almost creating an inverse of the curve displayed in the analyser.

4To deal with this, we turn up the Q parameter to make a very narrow EQ

band at 10kHz, then cut it by 10-12dB. Once you’re happy with your flattening EQ, you can add another EQ plugin for broad tonal shaping (for example, a boost around 2kHz often works well on guitars). The signal will also be much more receptive to treatment by other processors such as compressors and virtual amps/cabs.

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Thin. Harsh. Dull. Boomy. Nasal. Muddy. Tinny. No one wants their audio to sound like this, and the general cause of such problems is what we call a poor frequency response – that is, the relative levels of the frequencies in your signal.

There are many ways in which recorded audio can end up like this. Perhaps the instrument being recorded actually sounds that way. Or maybe you used a mic that picks up way too much (or not enough!) of a certain frequency range. The room you record in also greatly affects frequency response. Sometimes the effect can be very specific, affecting a narrow range of frequencies, causing “whistling”

resonances and ringing artifacts.It’s hard work mixing sounds

with disparate and inappropriate frequency responses, as you’ll have to apply unusual and unintuitive amounts of EQ, and dynamics effects like compressors may not react in the way you’d expect. It’s unlikely that you’ll achieve the best results working this way. We recommend a two-step solution: apply surgical EQ to correct deficiencies, then follow up with a separate EQ for the usual broad tonal shaping and mixing.

As you may know, EQ is short for equalisation, so-called because it was invented to even out the tonal character of a sound by

cutting and boosting certain frequencies across the spectrum, creating an even, balanced tone with each frequency at an ‘equal’ level. A good parametric EQ is an absolute must-have in any producer’s toolbox, and almost all of these feature a graphic display of frequency laid out horizontally across the centre, against gain or volume up the left and right edges. Several frequency bands can then each be centred around a specific frequency and adjusted to make that particular frequency louder (boost) or quieter (cut). A third control, mysteriously called ‘Q’, sets the width of the range of frequencies affected when you cut

or boost that particular band. Each adjustment you make to a band is reflected by a curve on the graphic display, so it’s easy to tell visually what effect your changes are having on the sound you’re EQ’ing.

Filters, meanwhile, are designed to reduce frequencies above or below a set point – the rate at which they do this is known as roll-off. A 24dB-per-octave roll-off is a much steeper ‘cliff’ than a 12dB-per-octave. High-pass filters (so-called because they only allow higher frequencies to pass through them) set at about 60-100Hz are useful for getting rid of rumble, and low-pass filters set between 9-10kHz are good for slicing off top-end hiss.

Frequency response issues

TUTORIAL FILES

In our _STEMS folder, you’ll find a badly recorded bass – the highs aren’t

crisp and there’s hardly any low end! Rather than correctively processing the signal, we’re going to use a synth bass to supply the missing frequencies. Set your tempo to 144BPM, bring up DuneCM on a MIDI track in your DAW, and load the patch 1. DuneCM Bass.fxp from Tutorial Files.

1This is a simple mono bass sound comprising sine waves an octave

apart and a tight amp envelope. Playing it back in the octave of C-1 alongside our recorded bass restores its beefy low end. You could use your DAW’s audio-to-MIDI function to get the part’s notes and timing, but for a simple line like this, it might be quicker to just sequence it by hand.

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This give us a solid low end, but the highs still need work. Duplicate the

bass track, and add IIEQ Pro CM to the new version. Put the first band in HPF24 mode, at 2kHz or so, to isolate the very highest frequencies of the signal, which can be distorted to generate and/or emphasise the lacking upper harmonics.

3We want the distortion to have a consistent tone, so we put CM-Comp

87 after the EQ, setting its Threshold to -56dB, Ratio to 2.0:1, and Makeup to 21dB. Then we add a distortion effect to grunge up the signal. For a more dynamic, lively sound, leave out the compression. (Audio: 4. Full frequency bass.wav).

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> Step by step 3. Fattening a thin bass sound with a synth Perfect match

We’ve shown you how to use EQ to flatten a guitar, but you don’t have to aim for a flat curve – indeed, the guitar (or other instrument) tones on your favourite records may be far from flat. You can use spectral analysis on these “golden tones” to see their frequency response, then apply EQ to get your tracks’ closer to theirs. Simply find and loop a part in your reference track where the guitar plays on its own, insert an analyser plugin (with the same settings as used on your guitar), then EQ your guitar until the curves look similar. Mid/side processing on the reference track can help – for example, on many rock tracks, turning down the mid signal (Voxengo’s free MSED is perfect for this) can give you an almost perfectly isolated guitar part. This matching technique is also useful with multiple guitar parts that you want to sound similar.

Alternatively, you can use a matching EQ such as iZotope’s Ozone 5 (being aware that its Source and Target labels are the wrong way round!) or Logic’s Match EQ to do it for you. These enable you to “capture” the tonal characteristics of a signal and automatically equalise your own audio to give it a similar sound, at far greater resolution than is possible doing it by hand.

Finally, understand that even the most accurate EQ matching will hardly ever make one signal sound exactly like another – but it will get you a bit closer!

> Step by step 2. Using notch EQ to eliminate mains hum

Mains hum can occur when recording external electronic kit such as synths,

electric guitars and turntables. In the UK, mains hum occurs at 50Hz, while in the US it’s 60Hz, both with harmonics. Solo the Synth track (from the STEMS_ folder) and you’ll hear a loud low-end hum. Add a gate plugin and set its sidechain filter to focus on the 500Hz range. Now, the synth will open the gate but the hum won’t.

1Set the gate’s Threshold to -50dB and Release to 170ms. Now, when

the synth isn’t playing, the signal will be silenced so you don’t hear the hum on its own. However, we can still hear the hum when the synth is playing, which will really cloud the bottom end of the mix. We can use EQ to notch it out, though. Add DDMF IIEQ Pro CM after the gate. (Audio: 2. Gated synth.wav)

2Set all bands to notch mode, and set them to 50Hz and its harmonics:

100Hz, 150Hz, 200Hz, etc. Maximise the Q on all bands, then adjust the Q of band 1 to taste. This takes out most of the hum. It’ll also adversely affect the low end of the synth part, but you can use a low shelf boost to help offset this if needed. A 24dB high-pass filter at 86Hz kills any remaining rumble. (Audio: 3. EQed synth.wav)

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September 2010 / COMPUTER MUSIC / 37



> Step by step 4. Creating double-tracked parts from one take

Huge, 3D-sounding parts can be made by recording the same thing multiple

times, known as double-tracking (or triple-tracking, etc). If you’ve only recorded a single part, though, don’t despair, as creating a double-track from an existing part is possible, provided there’s enough source material in the original track. We’ll start with the guitar part from the first tutorial, with our corrective EQ in place.

1Remember, our track has been recorded at 144BPM, so set your

project’s tempo to this. Duplicate the guitar track, then delete the audio that’s on the new track. Find a section where a guitar phrase/riff is played twice in a row and cut it into two halves – between bars 10 and 18 works for us, with the cut at bar 14. Make each half a different colour so that you can tell them apart.

2Copy the two sections onto the second track – this is where we’ll

assemble our fake ‘doubled’ track, using the first track as the ‘master’. Pan the first track hard left and the second hard right. If we hit play, the guitar will sound louder, still centrally panned and totally mono, as you may have predicted. That’s because the exact same part of the recording is playing back on both channels.

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To get a proper double-tracked effect, we simply swap the two clips over on

the second track so that the same two sections of the recording never happen at once – they now swap over between channels after four bars. Now we have got the illusion of two guitarists playing at once – one in the left speaker, one in the right, giving that big, wide, double-tracked rock sound.

4Work through the song using the same technique wherever possible.

There may be areas where you need to get creative to make the second track. For instance, when the guitar comes in at the start of the track, it seems there’s no corresponding part to copy to the second track; but because the lick is repeated later in the tune, we can just copy and paste one of those instances into place.

5Occasionally you’ll come across an unwanted variation, where the aligned

parts don’t play the same thing. For example, in this segment, there are different phrases at the end of bar 25, and a mistake in bars 18-20 of the right hand channel. Let’s first tackle the mistake. We zoom in and snip out the offending portion, then copy and paste a well-played phrase from bars 22-24 in its place.

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Joins and edits can be noticeable, so you may need to tweak region

boundaries to get patched regions to sit tightly. You can also use your DAW’s crossfade tools to smooth out transitions and kill clicks and pops. If you’ve patched up a section with mistakes, don’t forget to copy and paste the corrected version back into the place it came from on the original track to correct the original mistake, etc.

7To fix the variation at the end of bar 25, we use the same technique as

before, pinching a bar containing a ‘stop’ from bar 37 and pasting it in onto the right channel at bar 25, so that the two sides are now playing the same guitar part. Matching up the parts where they occur the other way round at bar 33 is trickier, as the left-channel part is only played once in the song. We’ll have to get creative…

8To recreate the necessary part, first set snap to 1/8, then copy and paste

the chord from 33.4.1.0 to 33.3.1.0, and the chord from 33.2.3.0 to 33.3.3.0. Use the technique described in step 7 if you need to tidy it up further. Not only do our guitars now sound bigger and more impressive, but they’ve been pushed to the sides, making space for the drums, bass and vocals to sit centrally.

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> Step by step 5. Using multiband dynamics to tame resonances in a vocal

Recording audio in environments that are less than acoustically perfect

can have the effect of boosting certain frequencies and suppressing others. Our vocal has been recorded in a front room rather than a recording studio, and using BlueCat Audio FreqAnalyst to examine it, we can see unwanted ‘spikes’ at certain frequencies, particularly around 1kHz, where things seem decidedly erratic.

1We could use EQ to lower these peaks, but this process would be

difficult to get right and could have a detrimental effect on the timbre of the vocal. A more natural-sounding solution is to use a multiband compressor to control the level of just the problem frequencies. Insert MMultiBandDynamics (demo at www.meldaproduction.com) on the vocal track.

2Select the 3-band compressor preset from the list on the left. To access

the plugin’s parameters, click the Edit button at the top of the interface. The graphical display in the top left hand corner shows us the preset’s three bands. Click Band 2 (the central one) to bring up its parameters.

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Drag the left and right edges of the band to set its frequency range. Set

the lower crossover to 700Hz and the higher one to 2kHz. This focuses the band on the range that’s giving us trouble. Click the Link button at the top of the Band 2 panel so that we can adjust this region’s parameters without affecting the others. Click the S button on the band to hear how it sounds soloed.

4By default the Threshold is set to -30dB, so we can reduce the area’s

dynamic range by simply turning up the Ratio. Move FreqAnalyst CM to the insert slot after MMultiBandDynamics to see the effect on the frequency response, and gradually turn up the Ratio. At 2.00:1 or so, the peaks are less jumpy. Turn the band’s Output gain up to 2dB. (Audio: Multiband compressed vocal.wav)

5As we’ve unlinked the bands, we can adjust the Ratio of Bands 1 and 3

separately, to apply gentler compression to them. We can now mix the vocal in the usual way. For instance, we add some width with MHarmonizerCM – turning up the Width parameter in the Global Pitch panel slightly – 0.35 semitones or so gives us a pretty sweet widening effect. (Audio: Stereo vocal.wav)

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Multiband processors split the signal into multiple frequency bands at user-defined crossover points. For example, setting the crossover of a 2-band processor to 1kHz will give us one band covering everything below 1kHz and another for all frequencies above 1kHz. So in a multiband compressor – the most ubiquitous of multiband effects – we’d be able to apply, say, heavy compression to the lower band, and only a little up top.

Most DAWs have multiband dynamics processors built in, and there are plenty of feature-packed third-party plugins available too. MeldaProduction deserve special mention here, as they offer by

far the most extensive array of multiband plugins on the market.

Some engineers avoid multiband processing for mix work, saying that if you have a good mix, they’re just not necessary. That’s all well and good if you only ever get perfectly recorded material to work with, but for the rest of us, multiband processing can be hugely useful for controlling unpredictable resonances, getting a grip on signals with constantly shifting tonal balance, adding targeted processing to mixed sound sources, and much more.

Take, for example, a vocal that’s been tracked in a living room. The shape of the room and its

acoustic properties will mean that it reflects some frequencies more than others, resulting in unwanted peaks at certain points in the performance. While this might not perhaps be all that obvious to the untrained ear – we’re used to hearing sounds in a natural environment after all – it’ll be clearly visible in a spectral analyser, and will make the vocal hard to mix.

The obvious solution is to use an EQ plugin to notch out the offending peaks, but while this works fine on some sources, it’s not really viable in this case, as the behaviour of the stand-out harmonics is just too erratic and varies greatly from note to note.

A more elegant way to deal with unpredictable resonances is to use multiband compression. This enables us to reduce the dynamic range of a specific range within the frequency spectrum, reducing volume spikes while maintaining the overall timbre of the sound.

Closely related to the multiband compressor, the dynamic equaliser (a great example being Voxengo’s GlissEQ, the demo of which can be downloaded from www.voxengo.com) responds to incoming volume level, increasing or reducing the amount of cut or boost applied to each band depending on whether the signal level hits a specified threshold.

Multiband processing

POWER TIP

>Variable GR Some EQ’s let you control the overall strength of the EQ curve. In EQuality, for instance, there’s the Range slider, which allows you to increase the maximum possible gain reduction of all bands, from the default -18dB up to a huge -36dB, or right down to 0dB, where cuts/boosts have no effect. Once you’ve located the troublesome resonant frequencies and got your curve in place, you can set the EQ to be as extreme or as subtle as you like – useful if your over-aggressive notching has robbed the instrument of some of its tone.

Be aware of potential phase issues when using multi-mic’ed drum parts



> Step by step 6. Fixing resonance and hi-hat bleed on a snare track

You’ll find our drum tracks in the STEMS_ folder – they’re all at 144BPM.

The snare in our drum track is quite ringy and resonant, and its mic has picked up a lot of hi-hat bleed. Let’s see if we can alleviate these problems. Solo the snare track and insert an EQ – we’re using EQuality again, with its built-in analyser enabled. Practically any EQ and analyser combination will work for this, though.

1To locate the troublesome resonances, we use the tried and tested “seek and

destroy” method. Simply increase the gain of one band and set a high Q, so that it boosts a very narrow frequency range. Play the snare on a loop and sweep the frequency through the spectrum until the main resonant pitch of the snare becomes really loud, ringing as much as possible – here, this happens at 482Hz.

2Next, use the Gain control to pull the offending frequency down in level

until it’s no longer bothersome. Now we add another band, repeating step 2 to find the next harmonic a little further up the curve. We repeat the process until we’re satisfied with the results. While we’re EQ’ing, we can also make the snare punchier by boosting its fundamental frequency – in this case, it’s around 225Hz.

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Hi-hat bleed is harder to deal with, but one solution is to use a denoiser

plugin capable of learning the noise profile of a sound for removal – we’re using iZotope’s RX2 DeNoiser. We isolate a couple of single hi-hats in the snare track from bars 4 and 5, trim off any adjacent cymbal or snare hits, then copy and repeat them to create a four-bar strip of hi-hats that we can use to feed the denoiser.

4We insert RX2 DeNoiser on the snare track, enable Learn mode and feed

it a few seconds of hi-hat. Once the plugin has learned the noise profile, it reduces those frequencies on subsequent playback. This kind of processing is always a compromise, so tweak the settings to find something suitable for your mix. If the snare sounds too dull, try inserting a second EQ and boosting the 5-6kHz range.

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The more mics you use to record a drum kit, the higher the likelihood that you’ll run into phase problems. This is because any number of the signals picked up by all those mics could well be out of phase with each other, leading to a thin-sounding drum kit with an overall lack of impact.

Getting a good phase relationship between mics is an essential part of obtaining a great drum sound, and is usually achieved by moving the mics around and checking and adjusting the phase before recording. If you discover phase problems after recording, however, the easiest remedy is to try flipping the phase

of individual tracks in your DAW. If you know two tracks are out of phase, simply flip one so that they work together to reinforce the sound, rather than cancelling each other out. You can zoom right in on the waveform to view whether the waves are in positive or negative phase at any given point.

If you’ve checked the phase and your drums still sound thin, another technique is to manually align the room or overhead mic tracks with the kick and snare tracks by nudging them backwards in your DAW by a few milliseconds, shortening the natural delay caused by the sound having to travel the extra distance to the

mic. This can work wonders in terms of tightening the sound, but the technique does run the risk of messing with the natural phase relationships between the tracks. If you do decide to try it, then, proceed with caution!

Another potential problem with multiple drum kit mics is crosstalk, or ‘bleed’, where the sounds of neighbouring kit pieces are picked up by each mic – so you end up with loud hi-hats in the snare mic, for example. The traditional approach to taming this has been to use noise gates, which can be tricky, but the current generation of dynamic noise reduction plugins can be useful in this area too.

The problem with drums



> Step by step 7. Fixing the timing of multitrack drums with Logic Pro’s Flex Time

Logic Pro 9 and X feature powerful timing correction in the form of Flex

Time. Let’s use it to correct the timing of a dodgy drum track. (If the rest of your parts have been played to fit the timing of the drums, you’ll probably need to adjust the timing of those too, otherwise they’ll end up out of sync with the corrected drums.) The project tempo has to be set correctly before we start, so set it to 144BPM now.

1Drag the drum parts into the arrangement, telling Logic to create

new tracks for each part. Delete any unused audio tracks. For proper multitrack editing of these drum tracks, we need to group them. To do this, press X to bring up the mixer, then hold the Alt key and click each of the drum track’s Group slots. This will set them all to Group 1.

2Bring up the Group settings by right-clicking a Group slot and selecting

Open Group Settings. Click the Editing (Selection) checkbox to tell Logic that we want to edit all the parts in this Group as a single entity. Keep the Phase-Locked Audio checkbox selected – this ensures that audio across the tracks stays in phase, which is crucial for preserving punch and coherence in multitracked drums.

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Close the Group Settings menu, and click the Flex Time button at the top

of the arrange window to activate Flex Time. By default each track’s Flex Time modes is set to Monophonic. This isn’t ideal for drums so change it to Slicing, which better preserves transients. As we’ve told Logic we want to edit all the tracks together, you only have to do this for one track and the others will follow.

4Double-click the kick drum track to open its audio editor, then click the

File button to view the waveform. From the Audio File menu, select Detect Transients. A series of orange transient detection markers will appear. Check the results by scrolling along and making sure all the kicks have been detected – it seems to have worked!

5Do the same for the snare track. You’ll notice that not all of the snares are

detected this time, and thus don’t have transient markers. You can use the + and – buttons to adjust the sensitivity of the transient detection, but a more accurate way is to hold Alt and click the undetected hits to add markers manually. Use the Eraser Tool to delete unwanted markers.

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Before using Flex Time quantisation, we need to tell Logic which tracks we

want to use as references for the timing of the other tracks. Generally, the kick and snare are the core of any drum beat, so we deselect the green Q buttons in the Track List on every track apart from those two.

7If you turn on the metronome, you’ll hear that the tempo of the drum track

drifts slightly. We can remedy this by selecting the kick track and setting its Quantize to 1/16th Note in the Inspector. This improves the timing, particularly in bar 7. (Audio: 8. Bar 7 quantised.wav)

8The timing of some of the fills might suffer from the quantisation, but this

can be fixed by zooming in and dragging the hits into the correct positions. For example, at the start of bar 10 the kick with the crash on it has been moved just before the beat by the quantisation. Simply drag the kick back onto the start of bar 10, and the rest of the drums will move with it. (Audio: 9. Fill timing fixed.wav)

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POWER TIP

>Overhead fixingWith our parts being triggered by MIDI regions, it’s possible to fix dodgy timing by quantising them. If you do this, however, remember that even if you mute the kick and snare mics, those drums may still be heard on the overheads, potentially creating flamming artifacts. This can be alleviated by editing the overhead tracks: group them together, generate transient hitpoints, then use your DAW’s audio quantise feature, set to the same grid setting as your MIDI part, to line them up. Alternatively, quantise only the offending hits.



> Step by step 8. Replacing drums with high-quality triggered samples

You can augment or replace recorded drums with high-quality samples. This

is called drum triggering, and many DAWs offer it, making it easy to trigger drum samples from audio tracks. We’re using Cubase. With our drum tracks loaded and the project at 144BPM, load the instrument you want to provide the new sound (BFD Eco in this example) on a new MIDI instrument track.

1Double-click the kick drum audio region to open it in the sample editor.

On the left, click the Hitpoints tab, then turn up the Threshold setting to generate vertical lines wherever Cubase detects a kick drum hit. Use the horizontal lines as a guide – you’re aiming to get the peaks of all the required hits to cross the horizontal threshold bar.

2Adjust the threshold until every kick drum hit has a vertical line just to the

left of it. If necessary, you can manually add or remove hit points by clicking Edit Hitpoints and Alt+clicking to do the deed. Now click the Create MIDI Notes option at the bottom of the Hitpoints panel. This brings up a set of options with which to fine-tune the process.

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You can choose to generate notes at a fixed velocity, but if you want to

preserve the dynamics of the original part, leave Dynamic Velocity engaged for now, as we have here. Leave the Pitch at C1 (the standard note for triggering a MIDI kick drum), and set the MIDI note Length parameter to a short value – we’ve chosen 1/32. Finally, choose the First Selected Track option, then click OK.

4The resulting MIDI region appears on our BFD Eco track. To keep things

simple, we’ll use a separate instance of BFD Eco for each piece of the kit (kick, snare, etc), so on this track we can mute the other drums giving us just the sound of the kick drum, without any bleed from the other pieces. If you can unload the other drums altogether, then even better, as you’ll conserve RAM and CPU cycles.

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Because we chose Dynamic Velocity mode in step 4, the volume of each

beat has been converted to note velocity; however, all the hits are too quiet. Open the MIDI part in the piano roll editor and adjust the velocities as you see fit – we select all the notes then drag their velocities up to the top of the velocity graph, so that the hits are loud but still have some variation.

6The beauty of triggering is that you’re free to change the drum sound by

simply loading a new sample into your virtual drum kit instrument. Put a section into loop play mode and flick through the available samples until you find one that suits the track. You can then adjust the levels of the mic’ed and triggered kick to get the perfect balance, or simply mute the mic altogether, as is quite common.

7Here we have a triggered snare drum part, created in the same way as our

kick. You can blend in the mic’ed snare if you like, to help maintain dynamic feel and realism. If you hear phasing artefacts between the mic’ed and triggered sound, try nudging the MIDI track forwards or backwards in time by a few milliseconds until it goes away. Our audio examples are in the Tutorial Files folder.

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> Step by step 9. Patching up dropouts and other unusable sections of audio

Load up the vocal track from STEMS_ in a project at 144BPM. Check out the

vocal track and take a listen to bar 50. There’s a potentially disastrous problem: a technical issue has resulted in a silent gap in the vocal! We need to find something to fill it, so let’s scour the vocal for a section with another sustained note at the same pitch. (Audio: 1. Vocal gap.wav)

1Luckily there’s something suitable at the end of bar 52. Cut out the missing

section at bar 50 and delete it, then slice out a section of the sustained note at the end of bar 52 and copy it to a new track, at the point where the vocal cuts out. Make sure to route this new track to the same destination as the main vocal track, eg, to a vocal processing bus.

2While the pitch is constant, the joins between sections of vocal are obvious

because of the clicks where the audio stops and starts. We can fix this with the judicious application of fades at the start and end of the repaired section. With the fades set up to keep a consistent volume level as shown, the join should be pretty much impossible to detect. (Audio: 3. Fixed vocal.wav)

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The term “audio restoration” brings to mind tedious and unsexy processes such as repairing wax cylinder recordings of Thomas Edison’s dog barking. Despite the negative connotation of its name, audio restoration software is wholly relevant to today’s musician, with the ability to salvage dodgy recordings.

Say you have a take that’s perfectly performed, but has been recorded too hot and as a result is clipped in places. Once upon a time, you’d have to simply accept this and move on – but not any more! Thanks to plugins like iZotope RX (version 3 of which was released as we were going to press), you can simply run the audio file through a “declipper” process to automatically restore the missing waveform tops, resulting in a perfect signal. You can download the demo of RX3 at www.izotope.com.

Feel the noiseAn issue that often comes up with recordings taken from tape is hiss, which in the past could only be remedied with EQ. Even in today’s tapeless studio, background noise can creep in from various sources, and the problem with broadband noise is that it covers a relatively wide frequency range, so the necessary EQ will usually have a detrimental effect on the signal as well. Thankfully, many audio restoration plugins and audio editors now have facilities to deal with constant background noise, including freeware favourite Audacity (get it at audacity.sourceforge.net).

In Audacity (the procedure is similar in most alternatives), highlight a section of the recording that features just the noise and none of the signal, select Effects»Noise Reduction and click Get Noise Profile. Audacity analyses the noise, and you can then use the various sensitivity and frequency parameters to tailor the level of noise reduction for the best possible result. Once you’re done, click the OK button to process the waveform, leaving you with a hopefully hiss-free version.

Restoration software

The great thing about these ‘learning’ noise reduction systems is that you can use them to remove any undesirable signal elements, particularly those that are constant in nature, such as mains hum or fan noise. It can even work on musical sounds, as we demonstrated on p41 with our hi-hat bleed.

Record breakersIf you’re taking samples from vinyl records, then you’re almost certainly going to end up with pops and clicks in your recording. While it’s possible to remove these unwanted glitches manually by zooming in on the waveform in an audio editor and using a pen tool to redraw it, this can be an extremely long-winded process if you’re dealing with anything more than a very short sample. Those in a hurry can

take advantage of the pop and click removal functions offered by software such as iZotope RX and Adobe Audition. You can download a demo of the latter from www.adobe.com.

Another problem that you might encounter when dealing with tracks recorded on malfunctioning or poor-quality kit is DC offset, where the horizontal centre of the waveform is offset from the zero line, resulting in a variety of issues, including causing compressors and distortion plugins to respond strangely. If you have a recording that appears to have a DC offset problem (it’ll be visible in your audio editor), simply apply your editor or DAW’s DC offset correction process to automatically realign the waveform. Alternatively, applying a high-pass filter set to its lowest possible frequency should also correct the issue.

iZotope RX3 offers powerful spectral editing and restoration effects that make light of hum, noise, clicks and more

One thing that we haven’t touched upon in this article is how to deal with situations where you’ve wound up with audio that is actually a recording of more than one instrument. For instance, guitar and vocals, or piano and percussion, etc.

Aside from applying the corrective techniques described elsewhere in this article, you may need to find creative ways of applying targeted processing to the elements of that recording. The most common problem is that the instruments are

not balanced in level, so first try some fairly quick compression on the entire signal to see if you can pull down the louder instrument’s peaks to make it seem quieter.

If that doesn’t work, instead turn to multiband dynamics, to control overbearing elements in certain frequency ranges and bring up low levels in others. For instance, if you have a combined guitar and shaker part, and the shaker is too loud, try quick, high-ratio compression on the upper (shaker) frequencies to sharply pull its level down on

each hit, and on the lower frequencies, gentler compression to boost the guitar’s musical tones.

Another possible route is to duplicate your source audio track and use filtering and gating tricks to isolate elements of the signal on each of the tracks. In our example, you could use filters to focus on the shaker, then set up a gate (or edit the audio by hand) so that it only opens on each shake. Now you can do things like send the isolated shaker to a reverb to give it its own sense of space.

Mixed signals

The Butterworth Band Shelf filter in DDMF’s IIEQ Pro CM is a fantastic harshness-busting tool, and the built-in analyser of the commercial version – shown here – makes fixing up your frequencies even easier

Throughout our tutorials, we’ve worked our way through practically every element of our demo mix, knocking audio gremlins on the head left, right and centre. With that done, it’s time to move on to the mixing stage; however, when dealing with low-grade source material like this, a few special considerations have to be made.

Keep ’em separatedAs we stressed earlier in this article, it makes sense to tackle problem audio in two stages: first, apply plugins and editing processes that will correct for obvious deficiencies and problems with the sound; second, go about your normal mixing routine of using sweetening EQ, phattening compression, sumptuous reverb, and so on, adding any such plugins after the corrective ones. Naturally, there is going to be some back and forth between the two stages – for instance, applying mix compression to the snare drum may make any annoying resonances even more audible, so you may want to go back to your notching EQ and pull down some of the cuts a little more. Or conversely, you might find that in the context of a mix, the twangy ringing of the snare adds character and helps it to cut

through, so you may prefer to back off the corrective EQ.

What’s vital is that you can separate these processes, to make the distinction between corrective and creative processing. For instance, if you prefer the sound of compression before EQ when mixing, don’t make the mistake of putting the compressor before your corrective EQ. If you do, it will then react to the highly unbalanced rogue elements in the original, uncorrected signal.

In the mixIn our Tutorial Files folder, you’ll find our final mixdown of our demo project, so let’s quickly run through our mixing procedure for this tune. The synth track needed no more work – we simply treated it to a little tape saturation from u-he’s Satin processor.

When working on our vocal, however, further ‘pokey’ resonances became apparent, this time in a narrow band centred around 2150Hz. To attenuate these harsh frequencies, we used one of our secret weapons: DDMF IIEQ Pro CM’s Butterworth Band Shelf with the Order setting turned up. This unusual flat-topped filter shape is extremely useful for busting clusters of harsh frequencies, as its square

shape means it will not dull neighbouring frequencies. Try it yourself! We then applied de-essing, our multiband compression, some gentle shaping EQ, the MHarmonizerCM widening effect, u-he’s Satin, and finally a slapback pingpong delay.

Drums were simple enough: we replaced the kick and snare with EZdrummer equivalents, applied our snare-resonance busting techniques to the toms, then sent all the drum channels to their own bus to be phattened up by Slate Digital’s Virtual Tape Machines and Universal Audio’s FATSO for UAD-2. In the end, we applied quantising only to the drum fills, to preserve the groove during the other parts. We also used cut-and-paste editing to add extra cymbals to liven up the repetitive performance!

On to the guitars. We added gating with FabFilter Pro-G to these to tighten up the stops, then our corrective EQ followed by UA’s API 560 for tasty tonal shaping, cutting a little at 1kHz to make the upper-mid presence frequencies stand out. A touch of CM-COMP 87 compression and Slate VTM tape saturation rounded it off.

To make our bass tone more cohesive, we routed all three tracks (the original audio, the low end synth, and the ‘excited’ top end) to their own bus, where we ran the lot through Studio Devil’s Virtual Bass Amp, then standard EQ, compression and tape simulation. This made it sound less like three individual elements and more like one solid bass sound.

To add a unified sense of space, we used a reverb effect on an aux send channel, sending the synth, vocal, snare drum and guitars to it, with a little EQ on the reverb to roll off its lows and highs.

On the master bus, CM-EQUA 87 gives a gentle 2dB dip around 1kHz, with stronger shaping from Universal Audio’s Manley Massive Passive EQ plugin. For dynamic control, we opted for Slate Digital’s VTM and Virtual Buss Compressors, and finally, iZotope’s Ozone Maximizer for limiting.

While our track might still not be perfect, and the mix we’ve done here is a quick one purely for demonstration purposes, we’re sure you’ll agree that it’s a massive improvement over the raw tracks, now sounding quite rich and smooth. Job done!

The final mixdown



is not playing precisely with the click. So if you quantise only the drums, or just the instruments, you might kill the groove!

MAKE IT CLICKOur demo project was recorded to a metronome click, which helped our editing processes greatly. If your audio hasn’t been played to a click, though, you might want to spend some time building a tempo map before getting stuck into editing. Some DAWs have tools to help with this, or even features to build a map by analysing the audio.

MAKE ’EM BOUNCEWhen you’ve got your corrective processes and plugins in place, consider rendering/bouncing the track to a fresh audio file to cut down on plugin clutter and general confusion. You can then delete the original (keeping a backup in case you need to go back to it), or just mute it and disable the

Zynaptiq’s magical Unfilter is designed to remove the effects of filtering and generally wonky frequency responses

plugins, re-rendering if necessary.

TOOLS OF THE TRADEThere are some fantastic dedicated tools out there for restoring, rescuing and repairing dodgy audio. Zynaptiq’s Unveil and Unfilter allow you to remove reverb and dodgy frequency responses from signals respectively, while dedicated correction tools like Celemony Melodyne and Antares AutoTune can be used to adjust the pitch/timing of performances. KNOW YOUR LIMITSThere’s a limit to how far badly recorded tracks can be made to sound better, so don’t drive yourself crazy trying to make yours sound like your favourite commercial records. Indeed, sometimes it’s best to embrace the weirdness of your audio and make it a feature of the track.

PERFORMANCE IS EVERYTHINGIf you’ve got the opportunity to rerecord your parts with a better sound, then go for it! But don’t ever forget that a great performance with a shoddy sound is often preferable to a pristine recording of a mediocre play-through. A perfect example: the vocal on Christina Aguilera’s Beautiful was recorded as a rough demo take, complete with headphone spill. The raw emotion of that take could not be bettered by later studio efforts, so the producers went with the demo version.

RECOMMENDED READINGObviously, the best way to clean up your recordings is simply to get them right in the first place! To help you do this, we put together Computer Music Special 62: Home Recording – The Ultimate Guide. Packed with tutorials on everything from capturing drums, guitars and vocals, to turning your house into a recording studio and multitracking a live band on your iPad, it’s a must-have for anyone even vaguely serious about recording. It’s available in shops, at www.myfavouritemagazines.com and on Zinio and Apple Newsstand now.

PATCHING UP If there’s a word in your vocal that’s just plain wrong (ie, a wrong lyric or mispronounced word) and you don’t have any extra takes to comp from, try reconstructing the missing word using syllables taken from other words elsewhere in the song. Is the same phrase sung in the next chorus, for example? Your DAW has the editing power to copy, paste, tune and tweak the smallest fragments of audio, so it’s always worth a try. Bear in mind that many consonants (such as T, C, P, K, and S) are unpitched, so you don’t need to take tuning into account for these – they can be taken from anywhere they occur in the song, regardless of the note the word is sung at!

MATCH OF THE DAY If you don’t have a ‘matching’ EQ (one that superimposes an EQ curve from a good-sounding source onto a different one to make it sound similar), try using two spectrum analysers, one on the track you want to emulate, the other on the track you want to adjust, and using EQ to make the displays match up manually. Even better, some analysers have multiple inputs so that you can see curves from two sources overlaid on one display. Or if your analyser has a stereo mode, hard pan the two signals left and right and route them into the analyser. FEEL DENOISEIf you’re using a noise reduction plugin such as iZotope’s RX2 Denoiser on a sound, be sure to place it before any compressors in the plugin chain. This is because compressors raise the level of the noise floor relative to the required audio, not to mention making the noise floor move dynamically, which would make the job of any noise reduction processors placed afterward more difficult.

GLOBAL APPLICATIONSOur demo project throughout this article is a pop/rock track, but don’t let that stop you applying our techniques to other styles. For instance, the approach we used to flatten out our guitar sound in the first tutorial can also be used to big-up dubstep synths and the like.

TRIGGER HAPPYWe looked at drum triggering, but why not “trigger” other instruments? Most DAWs now have some kind of audio-to-MIDI conversion for pitched instruments, which could be a possible solution when faced with a part whose sound is otherwise beyond salvation. Simply convert the part to MIDI and use a high-quality virtual instrument instead. For converting polyphonic parts, you’ll have to look into Celemony’s Melodyne. Alternatively, get stuck in and reprogram the part by hand – time-consuming, but possibly less so than the work needed to make a badly played/recorded part work in a mix!

OFF THE GRIDAs we touched upon elsewhere, in a real recording, musicians generally play along to the drummer, even if the drummer themself

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Documents

Clean Up Your Recordings (CM 197)