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A single pair of multipattern condenser microphones can provide at least five distinct overhead drum miking choices with surprisingly different sonic characteristics.

To demonstrate these different sounds, we recorded a drum kit with a pair of KSM44A overhead microphones five different ways. The drums, the room, the preamps, cables, and composition remained the same. The only variable was the position of the microphones (and in one case, the polar pattern).

How different can a single pair of microphones, in the same room with the same drums, really sound? As you listen to the tracks below, consider the following:

Width of stereo field

Mono compatibility

Susceptibility to comb filtering / phase coherency problems

Amount of ambient or room sound captured

Ease of setup

XY/Coincident Pair

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The XY or coincident pair approach wins points for its guaranteed mono compatibility. Because the two capsules are so close together ("coincident"), sound waves from every drum and cymbal impinge upon the two mics' diaphragms nearly simultaneously, avoiding any risk of comb-filtering due to partial cancellation of out-of-phase waves when the two OH tracks are heard in mono.

A consequence of coincident positioning is that the resulting stereo field is narrow, even when the two OH tracks are panned hard L/R. This is a good choice for arrangements in which the drums play a supporting role, or when mono compatibility is critical.

Overhead microphone height is an important consideration in this and every technique. If the mics are very high, the capsules will point straight over the sides of the of the kit, essentially putting the entire instrument off-axis. This will result in an overly roomy sound.

Lowering the microphones, or using an angle less than 90o can result in a more direct sound, although the kick and snare will always be off-axis.

Regardless of height, I recommend mounting the XY array directly above the snare drum, to ensure that the snare is centered in the stereo image.

Mid-Side
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The M-S technique, like XY, ensures mono compatibility. And like XY, it produces a relatively narrow stereo spread.

A primary benefit is that the perceived size of the space can be changed at mix time, by varying the amount of "side" channel in the mix. This is often described as adjusting the "stereo width," but to my ear the drum kit doesn't get any wider. The high and low toms don't get farther apart. The room gets bigger, but the drums sound less direct.

The challenge of M-S is that one of the two microphones is not pointed at the drum kit. The figure-of-8 mic is positioned so the drums are in the mic's null point; the microphone hears only reflected sound. If you are recording drums in a small or untreated room, chances are this reflected sound is not what you want on your record.

But if the room does sound good, M-S is a great way to combine a dry, direct sound - the Mid mic - with as much room ambience as suits the rest of the production.

Recorderman

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The Recorderman technique is probably the newest and least-known technique here, although it is conceptually similar to the overhead configuration of Glyn Johns' more-famous approach.

It requires two Cardioid microphones. One is suspended 32" above the center of the snare drum, pointing straight down. The other is positioned near the drummer's right shoulder, pointing directly at the snare drum from 32″ away.

This technique requires that the kick drum is also equidistant from both microphones. This can be checked by taping the two ends of a length of string to the center of the snare and kick drums, respectively. Pinch the string together at the point where it touches the mic above the snare, so that both legs of the string are taut. Move that point of the string in an arc toward the drummer's right shoulder to find the perfect location of the second microphone - at a point where it, too, is 32" from the center of the snare, and equally distant from the kick as is the first microphone.

The two tracks are panned L and R, although not necessarily 100%. Extreme panning of Recorderman overhead tracks can leave a hole in the middle of the stereo field.

Mono compatibility is not guaranteed, except for the kick and snare.

Because the microphones are relatively low, they will hear primarily the drums themselves. Cymbals will be understated, as will the sound of the room.

This is therefore a great way to combat some of the problems that plague live drum recording in home or project studios:

Small, untreated rooms that don't sound great

Untrained drummers who bash their cymbals

Even in large, purpose-built studios, the Recorderman technique is a useful way to capture dry overhead tracks that can be supplemented with more-distant ambient microphones, should the arrangement call for a roomier sound.

ORTF

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With a stereo bar, ORTF becomes very easy to use. Once the two microphones are locked into position - with capsules 17cm apart, at 110o- the single stand can easily be raised, lowered, or angled to tune the array to best suit the size of the kit, the sound of the room, and the desired amount of ambience in the recorded track.

Due to the wide spread of the microphones, it is likely that most of the drum kit will be between them, off-axis. This approach, like mid-side, is less likely to succeed in a small or untreated space, because the microphones will mostly hear reflected sounds. Moving the array closer to the drum kit, whether from behind or over the top, will result in a drier sound.

The ORTF stereo image is wider than XY, MS, or Recorderman, but not unbelievably so. Perhaps because the microphone position simulates human ear position, the resulting stereo field seems very natural and realistic, and generally collapses to mono without significant problems.

AB/Spaced Pair

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The AB or Spaced Pair approach is among the simplest stereo techniques to understand, but it presents the most problems in practice.

Because the microphones are far apart, AB arrays are susceptible to comb filtering. The risk can be mitigated somewhat by keeping the two microphone capsules equidistant from the center of the snare drum - but even then, the mic placement should be auditioned (in mono) to reveal possible problems with the kick, toms, or cymbals.

Aside from phase-coherency challenges, AB is a powerful technique, with a large palette of sonic colors. By lowering the array, or angling the microphones in toward the center of the kit, room sound can be minimized. Conversely, there are numerous ways to use an AB pair to capture a roomier, indirect sound: by raising the microphones, moving them further out toward the sides of the kit, angling them to point straight down, or changing from Cardioid to a wider polar pattern.

AB arrays can produce a wide stereo image, simply because drums or cymbals very near one of the two microphones will be perceived to come from only that side of the stereo field. This can result in an unrealistically large-sounding drum kit, if the overhead tracks are panned 100% L/R.

Head to head Comparison

Here's how the techniques compare:

Drum Miking Cheat Sheet

To minimize room sound, lower the OH microphones and point them toward the center of the kit.

To maximize room sound, raise the OH microphones, angle them away from the drums, or use wider polar patterns.

To minimize cymbal sound, try Recorderman.

To ensure mono compatibility, use XY or MS.

For the widest possible stereo spread, use AB.

Avoid phase coherency problems with kick and snare by keeping them centered between the OH mics.

Dry overhead tracks can be optionally supplemented with room mics.

A pair of cardioid mics provides several distinct, viable OH sounds.

Multipattern LDCs provide even more options.

About Matt McGlynn

Matt McGlynn is the founder of RecordingHacks.com. The site contains a searchable archive of specifications and insider information for over 1000 microphones, both contemporary and vintage, as well as in-depth reviews, a microphone sale-price finder, and a frequency-graph overlay comparison tool.

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Five Techniques for Stereo Miking Drums | Shure Blog

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A single pair of multipattern condenser microphones can provide at least five distinct overhead drum miking choices with surprisingly different sonic characteristics. To demonstrate these different sounds, we recorded a drum kit with a pair of KSM44A overhead microphones five different ways. The drums, the room, the preamps, cables, and composition remained the same.

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Ever wondered how the Dap Kings got that drum sound? In this video tutorial we take you step by step, including gear and plugin choices to get that sound.

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The vocal is the most important thing to a hit song. It encompasses the melody, rhythm and structure of a track. If you took the vocals out of some of your favorite tracks, they would probably become a repetitive mass of kick, snare and the four chord trick. It's the vocal layered on top that gives the song soul and feel. That's not to say that you should get away with recording a melody, putting it on top and then just leaving it. As mixing engineers we have to be aware that however soulful the lyrics and performance are, there are many mixing tricks we can do to push the envelope even further. What follows are a few mixing tricks to try on your vocal tracks. They may not work on every track, but these vocal mixing tips can give an interesting edge to your vocal recordings.

You have probably heard these effects before in some of your favorite tracks. Maybe you have wondered about how to recreate them. I'm going to shed some light on some vocal tricks used in modern recordings, from simple delay effects to robotic vocoding.

1. The Megaphone Effect

First of all, lets start with the megaphone effect. We're trying to recreate the effect of a vocal as if it were being sung through a megaphone speaker. Megaphone speakers are of the lowest quality you can find in a speaker and therefore we can have a pretty interesting effect by putting a clean vocal through the same amount of EQ and distortion a megaphone creates.

Here is the clean vocal:

I actually found a megaphone and spoke into it to be sure of the accuracy of this part of the tutorial. I had a general idea of how to recreate it and then just to be sure, I borrowed one and had a listening test. Sure enough, the final sound is very accurate to a real megaphone.

Because the megaphone only generates a narrow frequency range we start by inserting an EQ plugin on our desired vocal track. I filter out most of the high and low frequencies and then put a boost on the mids for that boxy and nasal sound.

Then to give the voice that little distortion we need, I inserted Logic's amp plugin and put a little drive on it. I also chose a 1x12 speaker so it would try to sound smaller. But whatever DAW you are using, you can recreate the same thing with whatever amp simulator you have, or even just a normal gain or drive plugin.

Thus, we have created the megaphone effect:

2. 100 ms Delay Effect

Another widely used effect is a medium delay with only a little bit of feedback. It thickens up the voice in a different way than the 25 ms delay used in artificial double tracking and gives it a tiny bit of effect as well. This track has a 125 ms delay which is actually in tempo with the track. It's not a huge effect because it sometimes gets drowned out by the rest of the track. But that's something you want with effects sometimes. An effect that's not noticeable unless you turn it off.

It's easy to accomplish this effect because the only thing you need is a delay plugin. Send your vocal track to a delay plugin and start your delay at 100 ms. Then dial in a little more or less depending on the feel and tempo of your song.

Be sure to have the sent delay effect underneath the main vocal track, as to not overpower the main element.

A basic mono delay plugin with a 20% feedback rate and a 125 ms delay. I added a little depth to the modulation section but that isn't crucial if your delay plugin doesn't have one.

So we've taken the dry vocal:

and added a basic but useful effect.

3. Creepy Whisper Effect

Sometimes you have a specific section that you want to enhance but you want to keep the main vocal relatively clean. There might be small section of a song that need an extra punch, or a specific ambience to the voice, but only for certain phrases or a few seconds. I want to show you how you can make a creepy whispering effect using, you guessed it, whispers. But we're going to effect the whispers so severely that they don't even sound human anymore.

This is the break in the song I want to enhance with some whispers.

If you have the chance during the production and recording, and have an idea like this, make your singer whisper some lines. It's good to have some extra recorded things to fool around with during mixing, and even though you don't know if you'll use, you'll at least get the chance to try. I have my singer double-track the vocal line in a whisper. Even dry, it's a pretty creepy whisper, it's closeness and creepiness an added effect in itself. But we're going to take it even further.

Listen to the whisper here. The tiny reverb ambience that you hear on the track was added in the recording phase. Don't worry, it's not crucial for the effect.

I'm going to double the effect, and have two variations of the same whisper. So let's copy the recording onto two separate tracks. On the first one we're going to accentuate the telephony, raspy quality of the whisper. So let's take out all the low and high end, cutting everything until 1 kHz. Then filter everything out from 2-3 kHz onwards. Now we're left with only an octave or so of voice, which makes the whisper all the more whisper-like. ake a bell curve EQ and spike a desired area between the two filters.

Compress the vocal so it squashes down at a high ratio so it stays level, consistent and powerful.

Now for the final creepiness factor. We're going to modulate the voice for an added horror effect. We want it to almost sting to listen to, so I'm slapping on a ring shifter. The ring shifter in Logic is a modulation effect that shifts frequencies around and alters the harmonic relationship of the original signal. With a little fiddling around we can come up with a severe enhancement to our whisper effect.

If you have a ring-shifter in your DAW try to copy these settings as well as you can. The most important things are which frequency you choose to shift, as going too low ends in a Darth Vader style shift, and going too high has a Chipmunk type effect. But we want to keep things creepy so we're just shifting the frequencies we still have after our EQ, or around 1 kHz. By skimming through the remaining frequencies you can find a sweet spot in your vocal track. Using the LFO section we can make a smooth frequency shift, as opposed to a more square wave one. As before, it adds more to the creepiness factor by having the shift as smooth as possible.

Listen to the first whisper effect here:

Now since we want to have an equal amount of creepiness on both sides of the stereo spectrum we're copying this whisper but putting a whole different type of effects on the other one. We're focusing more on the lower register, with a little vocal transforming, detuning and reverb.

Copy the whisper to a new track. This time we're not filtering anything out but using the whole tonal spectrum of the vocal. As before, compress the whisper pretty heavily to keep it consistent. Then insert a vocal transformer.

The vocal transformer in Logic let's you make subtle pitch changes to the voice, without changing the original value of the track. We're just putting a subtle Pitch and Formant Change on the voice, with the mix at around 50% or so. If you don't have Logic you can achieve a similar effect with a different type of plugin. Try using a subtle pitch shifter and enhance it with and Exciter or EQ, boosting the high frequencies.

Now insert a pitch shifter. We shifted up the whisper with the vocal transformer in the last step, so we got a more airy or breathy character to the sound. But now we're adding a pitch shifter and detuning the vocal down 5 semi tones. This way we keep the breathy quality of the vocal transformer (or exciter if you are using one) but with a detuned low end effect.

Keep the mix fairly low, or around 30-40%. We don't want it to sound harmonized, we want it to sound creepy. So we get a ghost-like detuned effect on the whisper.

This is what we've got so far.

For the final touch we're going to add some delay and reverb for ambiance.

I've synced this delay to the track and filtered out the low and high end, similar to what I did to the other whisper. Since I'm inserting this on the track as opposed to sending it like you normally do with delays, be aware that you need to keep your mix low so it won't overcrowd the original signal. We want to add a little bit of creepiness to the whisper, but we don't want grown men crying themselves to sleep over it. Finally, add a little reverb with your plugin of choice. Choose a small hall setting or a plate with a medium reverb time. Play with the dry and wet controls until you get just the right ambiance for the track.

Listen to the second whisper track:

Finally listen to it in context with
the main vocal, and the rest of the track.

4. Advanced Artificial Double Tracking

A.D.T. is a widely known mixing trick. It's used on anything that needs to be a little bigger, a little wider, a little better. It's the process of taking the main vocal and doubling it via a delay or a chorus. It's less expensive to double your singers track artificially than have him come in and waste precious studio time by singing the whole thing again. Thus, it's easier to just copy the vocal, process it a little bit, usually with a delay, and have an instantly doubled track without all the effort.

The normal double tracking is done with sending a vocal to an aux, adding a delay with about 20-30 ms and no repeats. That way we have the same vocal, note for note, but the subtle time shift makes the vocal sound much bigger. We're going to take things even further, sweetening it with more than just delay. We'll be adding slight pitch shifting as well as an exciter to add an extra layer or freshness.

Start by sending your vocal to two separate auxes. Insert a tape delay on both auxes, set to 23 ms without any feedback. Try to recreate the screenshot below in the DAW of your choosing.

By having no feedback we get the double tracking effect by using delay. If we were to have a little feedback we could get a type of slap echo, similar to the typical John Lennon vocal sound. Remember to have this on 100% wet since this is a send effect. Now repeat this on the second aux. You can add a different millisecond value if you want, but I'm using the same settings. Now pan both auxes 30% left and 30% right. We're widening it with manual panning as well.

Now we need to add a pitch shifter on both tracks. Since singers aren't robots they never sing with quite the same intonation and pitch. By adding a slight pitch shift to the vocal we get that extra nuance we need. Again, 100% wet and with a 12 cent pitch variation. On on aux I detune the vocal down -12 cents. On the other one I pitch it up +12 cents.

Now we should be getting pretty close to a sweet doubled vocal. The vocal you had before definitely sounds better now than before. But what if it still sounds kind of dull? What if we need just an extra bit of excitement to the dull sung vocal? Easy, let's pop an exciter on the auxes and give it some air.

Insert an exciter plugin on both aux sends. Exciters add artificial harmonics to the recorded tracks, giving them a little life in the treble department. So if you have a dull, lifeless recording you can try to see if an exciter can help you out. I have used it a few times on different instruments, guitar solos and vocals to name a few.

We're just going to add a little air, setting the exciter on a high frequency. I have mine on about 13 kHz, but you can experiement with a higher or lower frequency, depending on the type of track you are enhancing. Of course, add a second exciter on the other aux and vary the frequencies a little bit.

Now we have a thick, wide and deep double vocal. These simple ADT tricks have definitely transformed my vocal track, prompting me to think that I should maybe mix the song again.

We've turned this:

into this:

Incredible difference!

5. The Vocoder Effect

The vocoder, or VOice enCODER has been around for a while. Since the seventies it's been used in a wide variety of musical and cinematic projects. Prog rock and early electronica took an instant liking to the vocoder and it was used in many tracks from the seventies onwards to present day. Pink Floys, The Alan Parsons Projects, ELO and many more used the vocoder in some of their songs. Vocoders in modern music programs combine the classic voice encoder with a synthesizer. Effectively combining a polyphonic synthesizer with a voice encoder, enabling you to play along with the audio you want encoded. In modern music one of their more famous version of recent years is Imogen Heap's Hide & Seek.

As amazing as this song is, the good thing is that we can recreate this song pretty realistically using a few DAW tricks.

Vocoders work by listening to an incoming audio signal, in this case a vocal track. By combining this vocal track with the MIDI information we put onto the vocoder track we get this robotic vocoder effect. So we need a vocal track and a MIDI track. I am using the original piano track that was recorded as the MIDI information. Be aware that you will have to have MIDI information that is in key with the rest of the song, otherwise you might get undesirable results. The piano track in my example are just simple chords that are accompanying the vocal, and thus makes for a great example for our vocoder trick.

Take a listen to the original track.

Let's start by creating a software instrument track or a MIDI track. Depending on your DAW there are different ways for you to insert a vocoder. I'm using Logic and they have a pretty powerful Vocoder synth. In Reason the Vocoder works a little differently as you need to patch things yourself on the backside. Whatever vocoder you are using, there should be a simple way for you to get the hang of it. But lets focus on the one in Logic for this particular example. Insert it onto the empty software instrument track.

If you have a MIDI track that you want to use as the basis for the vocoder, copy it onto the vocoder track. This MIDI data is what will give us the pitches of the end result, but with the singing characteristic of the vocal track.

Insert your vocal on the sidechain of the vocoder. This way the vocoder listens to the vocal track and sings it back after processing. You will get the harmonic content of the piano chords with the articulations of the vocal. Now the only thing left to do is tweak your vocoder settings until you are happy with the results.

Now there are a few things to keep in mind when working with the vocoder. Be sure that your vocoder understands that your incoming signal is a voice by checking the Voc at the top right. Sometimes this isn't needed but as the EVOC 20 doubles as a synthesizer as well you are able to select what type of signal you are taking in.

Being that this is the voice and keyboard combo you want it to be in poly mode. That way it analyzes all the MIDI notes and plays them back at the same time, resulting in a more harmonically rich effect. You also want the side-chain analysis to be quite fast so it robotizes the voice faster. With a longer attack the voice seems to come through more. But that can also be an effect you are looking for. We're going for a full on effect, not a subtle mixing technique. You might want to filter out the lower frequencies as robotic bass doesn't really sound good. I have a cutoff at 130 Hz but you might want to go higher, depending on your track. Play along with the various knobs and parameters until you find a good sound. Use the screenshot above for a good starting point, but by tweaking you can get some radically different results. A bigger stereo width, a more fluctuating LFO or a different frequency modulation are all things you can experiment with.

Then you have to decide if you want it as an effect underneath both instruments, or if you want to have either one complimenting the vocoded effect.

Here's the effect underneath both the main voice and piano.

Here I've taken out the original vocal track and put the piano and vocoder together.

And here is the vocoded effect soloed.

Let's do one more thing and give the effect some ambiance and depth. Send the vocoder track to an aux buss. Insert a chorus and try for a shimmering wide setting. That's usually accomplished with a low rate and a high depth or intensity, like in the screenshot below.

And let's throw a reverb after the chorus filling out the space even more. We're getting close to the super produced Imogen Heap sound of Hide & Seek. Choose a medium to large hall setting for a bigger sound.

Listen to the final result, a big wide and shimmering robot singing effect.

Conclusion

Sometimes you struggle with trying to recreate effects heard on records without knowing where to start. You end up going through all your plugins in hope to find a preset that says "that-cool-vocal-effect" without any luck. At least you can tick all these vocal tricks off your list on things to learn, keeping them in mind when you want a special effect to go with your main vocal sound. Of course, a badly recorded vocal can't be salvaged by piling effects on top of it. The vocal has to have its special character before you start dressing it up. So be sure you have something to work with before you start enhancing it with effects.

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The vocal is the most important thing to a hit song. It encompasses the melody, rhythm and structure of a track. If you took the vocals out of some of your favorite tracks, they would probably become a repetitive mass of kick, snare and the four chord trick.

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Which plugin gave "Uptown Funk" a tougher sound? How many drum mics were used on Amy Winehouse's Back to Black? Find out in this exclusive interview.

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how to record drums with two microphones

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Recorderman Technique - The Real Story from the Recording Session it was Invented At

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Eric "Mixerman" Sarafin was there when the Recorderman Technique for recording drums was born. He gives us the whole story about how it happened. The big shocker: Close microphones on the kick and snare were used, and it was never intended to be a two-mic recording technique.

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How to record Heavy Guitar part 3-THE FREDMAN TECHNIQUE | TUTORIAL

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More Guitar Recording Tutorials HERE!: http://bit.ly/1YIxVhD Get a second SM57 here: http://amzn.to/1MCAhlD MERCH: http://www.spectremedia.ca/shop Patreon: http://www.patreon.com/smg Exploring the "Fredman Technqiue" of miking a guitar cabinet. If you're looking to add a new dimension to your tone, have a look at this, it'll open up a whole new way of recording metal guitar to you!

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Art of Audio Recording: Recording Guitars - 5. Multiple Mic Techniques

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Video 5 of 16 from macProVideo.com's (The) Art of Audio Recording: Recording Guitars For more information visit: http://www.macprovideo.com/tutorial/recording-gutiars?utm_source=YouTube&utm_medium=Social&utm_campaign=recording-gutiars The electric, acoustic and bass guitars are the foundation of most modern music. This tutorial teaches you everything you need to know to record guitar tracks that will jump out of your mix and blast your tunes to the top of the charts.

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Producer and engineer Rob Kirwan oversaw Hozier's commercial breakthrough - by concentrating on the least commercial aspects of his music.

"While working on Hozier's music, I had absolutely no idea of the impact it was going to have. Andrew is a lovely, honest and insanely intelligent man, and I really enjoyed working with him. He's extremely talented, and I thought he would have a slow burner of a career, with a gradual climb to international success, like a Nick Cave or PJ Harvey kind of artist. Even when I first heard 'Take Me To Church', I did not think that it would become a big radio hit. But the radio in Ireland just went nuts for it, even before the video was released. I never thought Andrew would have the instant success he is currently enjoying."

Engineer, mixer and producer Rob Kirwan was approached by Irish label Rubyworks in May 2013 to work on an EP by an unknown singer, songwriter and guitarist they had recently signed. Andrew Hozier-Byrne had studied music at Trinity College in Dublin, but dropped out to pursue a music career, and had been a member of the vocal group An na from 2008 to 2012, appearing as a soloist on one of the tracks on the group's album Illumination. Hozier-Byrne had no track record beyond that. Insofar as Kirwan was concerned, the request from Rubyworks concerned a typical low-budget development project.

"Andrew had worked with a few other producers," recalled Kirwan, "with quite poppy results, with which he was not very comfortable, so he never went forward with those recordings. So he set about doing some demos in his attic, on a Logic system, and the brief from the record company was for him to redo his vocals on three songs, and for me to then mix these tracks. When I heard the demos I felt that the vocals were the best thing about them, but that we had to re-record everything else, because many of the instruments sounded too thin and not powerful enough. So we did it exactly the other way round! They gave me a budget for four days in my studio, but when we started re-recording everything I realised that it was going to take much longer, so I kind of gave Andrew a few extra days for free. It took us eight to nine days to re-record and mix the three songs on the EP." (The fourth song that appeared on the EP, 'Cherry Wine', was a live recording in which Kirwan took no part.)

'Take Me To Church' was the title song of Hozier's first EP, and both EP and single were released in September 2013. The single reached the top of the Irish charts a few weeks after its release. Hozier's ear-catching blend of rock, soul, gospel, blues and Irish folk music and the immediately apparent integrity of his lyrics and music clearly hit a nerve, as did the controversial and arresting video, which portrayed homophobic violence against a young gay couple. According to an interview in Billboard Biz magazine with Rubyworks owner Niall Muckian, 15 major-label A&R men chased Hozier and Rubyworks within weeks of the single being posted on YouTube. Deals were subsequently signed with two major labels, Universal and Columbia/Sony (for the US).

Hozier began touring extensively, and with the chemistry between Kirwan and Hozier apparent on the first EP, the next step was a return to Kirwan's Dublin studio at the end of 2013 to make EP number two. "The budget and expectations had gone up enormously," recalls Kirwan. "All of a sudden it was clear to me that Andrew was no longer an unknown, local Irish artist, that he was going to become something far bigger than that. We recorded four songs for the second EP, but one of them wasn't included on it, because it was supposed to be a single, and they said, 'Why is it not all poppy and shiny?' Instead 'For Eden' became the lead track, and it was mixed by Mike Crossey. I mixed the other two songs." (The fourth song was once again a live track.)

'Take Me To Church' was released in the US in the first half of 2014, and proved a slow grower. With Hozier appearing on the David Letterman and the Ellen DeGeneres shows, various radio stations and festivals, the single gradually climbed up the charts, until it reached number two on the Billboard charts by the end of the year. The single reached the number-one spot in more than a dozen other countries, and ended up close to the top in a dozen more, including the UK. By the time of writing it had still not run its course. Kirwan and Hozier went back in the studio again in the beginning of 2014, to record the singer's debut album, which was released last September and became the fastest-selling Irish album of the year. It remained at the top spot for five weeks, and looks set to become Ireland's biggest-selling album for a decade. Hozier also reached the number two spot on the Billboard 200 albums chart and number five in the UK, and is still roaming around the top of many charts.

The enormous success of 'Take Me To Church' and Hozier has proved a career-changer for engineer/producer Rob Kirwan as well as for the artist. Kirwan played drums early in his musical career, but never had ambitions to become a musician. Instead he joined the legendary Windmill Lane Studios in Dublin in 1992 as a tea-boy, and worked his way up. Three and a half years later, U2 offered him a full-time position at their studio, where he worked with producers such as Brian Eno, Flood, Alan Moulder and Mark 'Spike' Stent. Kirwan moved to London in 1998, where he lived for 10 years, continuing to work with Flood and with acts like Depeche Mode, Erasure, Gary Numan, New Order and Editors. After a year of working and living in Berlin, Kirwan moved back to his native Dublin in 2009. He has since worked with PJ Harvey on her album Let England Shake, and recently also had major success in Ireland with the band Delorentos.

Almost all the recordings and all the mixing that Kirwan carried out for Hozier took place at his Exchequer studio in central Dublin, which he shares with Nick Seymour, Crowded House's bassist. Although Kirwan is proud to have received his studio training on analogue equipment, Exchequer is a desk- and tape-free zone. It nonetheless contains a large amount of unusual gear that Hozier and Kirwan put to good use. The latter explains: "I have a lot of stuff that makes bands scratch their heads and go: 'What's this? It doesn't sound like any record that we've heard!' Whereas Andrew would always go: 'That sounds amazing!' He liked all the crazy gear that I have been collecting, and what's more, he knew how to integrate it into his music. His was one of these records on which nine out 10 crazy ideas that came up actually worked.

"This meant that Hozier was a great record to work on for me. I love things that have character, and I love good performances. Energy and excitement are big things for me. Rock & roll music should sound exciting. The first record I ever bought was Motörhead's 'Ace Of Spades', which is probably one of the most exciting records you'll ever hear. That's the template for me, as well as my experiences at Windmill Lane. You had to line up the tape recorder there every day, clean the heads, and so on, and you saw how producers worked. If you come from that background, you understand sound better, and you understand that everything is centred around the musician and the performances of the musician.

"People today focus way too much on the computer, which they turn into the master of everything. But what I am interested in is the sound of people in a room, and how that affects you emotionally. You can do anything in a computer, apart from turn a boring performance into an exciting one. And you can certainly make an exciting performance sound boring in a computer! Chopping up drums in a computer so they all sound the same is not interesting. The computer's capabilities are endless, but you don't necessarily need them, and worse, they can be a major distraction. They invite you to forever postpone committing yourself, and this does not necessarily help music at all. I worked on a couple of records with Brian Eno early in my career, and one of his greatest attributes was that he always insisted on making decisions now. 'Does it sound good now? If so, then the decision is made, if not, we'll erase it and do it again until it sounds good.'"

The gear at Kirwan's studio that inspired Hozier included analogue goodies like the Inward Connections Vac Rac EQ, Teletronix LA2A, Anthony Demaria Labs 1000 compressor, Neve 33609/J reissue compressor, Manley Massive Passive, Neve and API mic pres, and a '70s Klein+Hummel mastering EQ. His main monitors are ProAc Studio 100s, with a Quad 405 amp. Kirwan bought much of his gear in Berlin, and notes, "I always like buying things that nobody else has, which makes me think that I am doing something slightly different."

A few items caught Hozier's attention in particular, said Kirwan. One was the latter's "Eastwood Airline guitar, which I bought after working on the PJ Harvey album [Let England Shake], because so many bands play run-of-the-mill guitars. Andrew loved it, and it played a very prominent role on the album. It was sent through a [Electro-Harmonix] Memory Man pedal, with the gain turned up loudly, a ProCo Rat guitar pedal for that distorted guitar sound, and one of these new hand-wired Vox AC15 amps, and then a Knas Ekdahl Moisturizer spring reverb. This was the guitar chain and it gave a really interesting tone that suited Andrew's music. I also used the Ekdahl reverb extensively on his vocals, as well as my Neumann CMV563 microphone with an M7 capsule and Martech MSS10 preamp. This signal chain distorted in a nice way, and Andrew adored that sound. He can sing really loud sometimes, which meant that we got a distortion similar to that from the 1950s and '60s.

"And finally, a lot of people, including artists, have been asking me for the nature of the reverb on Andrew's vocals. It's a combination of my Knas Moisturizer that I already mentioned, and the Altiverb [plug-in]. I used a Wendy Carlos EMT plate [preset], from which I removed all the top end, and to which I added a pre-delay."

All these character elements are present in 'Take Me To Church', adding greatly to its distinctive sound and undoubtedly playing a significant part in the song's enormous and instant impact. Rewinding all the way to the first moment Kirwan heard the song, in May 2013, he recalls: "I had no idea that it was earmarked to be the first single. In any case, I always try to treat every song I work on equally, whether I'm doing an EP or an album, and then the label and artist can later decide what song will be the single. If you tarnish a demo with the word 'single', everybody starts freaking out and becomes really precious, and it's more difficult to allow the song to develop. So in my mind I am always working on all songs equally, and then after I am done, the client can decide on the business side of things.

"In general, the songs that Andrew came in with were pretty complete. With 'Take Me To Church' it was purely a matter of sonically making it work. The arrangements were already there. Andrew is one of the most talented arrangers that I have ever come across. And he was 23 when we did this, as well! He is doing very well for a reason, which is his talent. If you look at the very top of the screen grab of the Pro Tools session, at the time-signature changes at the meter throughout the song [going from 4/4 to 6/8 to 8/8 to 6/8 to 4/4 and so on], that was all him. It took me ages to work out how to do it in Pro Tools, to be honest!

"'Take Me To Church' came pretty fully formed, apart from drums. We just needed to make the whole song sound bigger and wider and fatter and deeper and all those things. So we decided which elements of Andrew's Logic demo we wanted to rebroadcast [re-amp], and which we wanted to rerecord. The Hammond organ presets sounded a bit digital, so I would have broadcast that. But things like guitars, Andrew is such a great guitarist that I said to him: 'Look, that doesn't sound great, why don't you just play it again?' He'd go through the chain that I described to you earlier. Once again, it was very much a matter of improving the sounds, rather than the parts.

"I don't remember suggesting many structural changes to Andrew's other songs as a producer, because he's such a great songwriter. Maybe we did a few edits here and there, but by and large the songs were well developed. The arrangements for some songs were written in stone, and with other songs they were more open-ended. In a few cases it was different. With 'From Eden', he came in with just a guitar hook and the vocals, and we had to make that work on a completely different level. With 'To Be Alone', which is one of my favourite songs on the album, there was a lot of playing around with arrangements and trying out crazy sounds.

"Of course it helped when the budgets went up, because it allowed us to take more time. When we did the album, which took us three months, we did two or three versions of a couple of songs to keep the labels happy. In general it was Andrew and I working together, and then bringing in musicians as we needed them. The other difference between working on the first EP and the other material was that we recorded the drums for the first EP at my studio. Everything here is fully set up and integrated and ready to go, so I do 80 percent of the recordings of most albums here, but the acoustics are pretty dead and not great for recording drums.

"Normally I go to an outside studio a mile away to record drums, but because of the low budget I decided to do it in my own studio, which is why they sound the way they do. It took some smoke and mirrors to get the energy from them that I was after. The drums for the second EP were recorded at Cauldron Studios, and for the rest of the album at the place where I normally record them, Westland Studios. Oh, and the final difference is that Andrew Scheps mixed all the songs on the album, apart from 'Take Me To Church'. The American label also got Andrew to mix that song, and I loved his mix, because he also tries to connect with emotion first, and sonic second. I don't really care whether music is sonically beautiful or not, the most important thing is whether it touches me. Andrew [Scheps] is the same. In the end his mix of 'Take Me To Church' was not used because my mix was doing so well."

Rob Kirwan explains that some of the drum tracks on the 'Take Me To Church' screenshots are almost black with edits because, "there were some problems with the timing. This and the issues with the sound are the reasons why the drums aren't very prominent in the mix. I normally like drums to be right up there, as you can hear on the rest of the album.

"I used more or less the same recording setup for recording the drums at my place as I later used at Cauldron and Westland. I take all my mics and mic pres and other required gear with me, because they create my sound. Generally, particularly if I don't know a room, I will ask the drummer to come in with just a kick, snare and hats, and will get him to play in different parts of the room. I'll be looking for the place where I hear the most bottom end on the drums, and I'll also ask the drummer to tune the snare drum, so it sounds interesting and ballsy.

"My signal chains usually consist of an AKG D112 and a Neumann FET U47 on the kick, going through my Neve 1073, and I record the snare with a Shure SM57 on top, going through an API 3124+ mic pre, and a [Neumann] KM84 underneath going through whatever is available, often the desk at the studio I'm working at. The API can clip on the way into Pro Tools, so I generally control the level with a compressor. When you stick a pad on the mic, it never sounds good to me. My overhead mics are Gefell M58s, and they go through my Summit TPA 200B tube preamp, and then though a [Urei] 1178 with very little compression. The toms usually are Sennheiser MD421s, but I'll also have an additional 421 mic underneath the floor tom, and I'll flip the phase. This gives me the weight I want from the floor tom, combined with top-end attack. Other than that I could be using any of the unusual microphones I have or that the studio has. I have a couple of Placid Audio Copperphone mics that I may try, and they can sound great.

"I also put my Neumann CMV563 M7 up as an ambient mic, which goes into my API mic pre, which is really good for exciting top end with a lot of energy, and that into my Empirical Labs Fatso Junior compressor, and I compress the f out of it. This is my 'go to' drum sound. I want the compressed energy of the snare and kick drum, but I have to be careful not to have the hi-hats and cymbals too loud. The sound of my Fatso chain was a large part of the sound of the drums on the Hozier album. Apart from the Fatso track, I record clean as a whistle, with very little EQ or other treatments on the way in. I also play around with phase correlations a lot, but I don't have the patience to move microphones a quarter of an inch and then check the phase correlation each time. I just flip the phase on everything, and try all the different phase permutations that are possible, and listen to how that affects the overall sound. I love recording drums!"

"The bass guitar went through my '70s Ampeg SVT bass amp, on which I had a Neumann FET 47, which went into a 1073 mic pre, and then through my Teletronix LA2A. Another big thing on the record is my Sequential Circuits Pro One synth, which became a big part of the sound of the bottom end. I tend to record that DI. Nick [Seymour] has a beautiful Bösendorfer baby grand in the studio, and I used my two beautiful Gefell M58s on them, which have a lovely, warm sound, and I'll usually will stick them through my Summit mic pre and then the 1178 - as I do with the overheads. There are also strings on the album, which were done by two string players who came to my studio, and overdubbed using a Gefell M62 on the cello, and my Neumann KM84 on the violin. The preamps would have been any I had spare at that time.

"Guitars I generally record using an SM57 and a Sontronics Delta. The former will go through a 'lunchbox' Neve 1073 copy, and the latter through an API 3124+. I'll compress that with the ADL 100 and an LA2A. The Delta is a ribbon mic, and I use it a lot. I like Sontronics' ribbon microphones. They give you that ribbon sound, but the problem with a lot of traditional ribbon mics is that the gain is so low that you end up bringing up a lot of noise to get them to speak. I don't think it is worth it, whereas the Sontronics microphones require phantom power to start off with, which is pretty wild, and they are loud. They really complement the 57. The Delta mic gives the guitar lovely warmth and big bottom end, and the 57 picks up the edgy stuff in the middle. I use those two microphones on pretty much every guitar amplifier that I record.

"I recorded Andrew's vocals with him in the room, right behind me, so there were no natural acoustics on them. I already spoke about my CMV563/M7 vocal chain. I also always had a Sennheiser M441 on him as a sort of safeguard, and I combined these two while mixing. My mic pre on the 441 was the 1071, and I also had the Vac Rac EQ on his vocals, and the LA2A. So the vocal chain was as warm as it could be. The mic he used in his attic is a Neumann U87. The problem with 87s and 414s for me is that they are so accurate that everything sounds pretty sterile on them, hence my preference for using Gefells and other unusual microphones with lots of character. But it worked on his vocals because his performance was so good."

"I don't have a mixing desk, because as much as I like my analogue gear, people now want recalls in milliseconds. And that's impossible to do in the analogue domain. If one knob on an 1176 is a fraction out, the mix will sound entirely different. You simply can't mix on a desk any more. I don't care how good your assistant is, a desk recall never sounds the same. And stems also don't quite sound the same to me either. I use a lot of analogue gear on the way in, and I have the Chandler summing mixer to give more of an analogue feel to the mix, but I'm largely mixing in the box now, apart from the occasional outboard effect and things I send out and re-record back in again. The only thing I have is a PreSonus fader [controller] if I want to write in my levels using that. Other than that it's just mouse and keyboard.

"One of the advantages of working with a DAW is that it's easier to mix as I go, so my roughs sound very similar to the final mix, and I sometimes prefer them to the final mix. The great thing in working on a DAW is that you can always go back to an earlier mix, if you prefer the feeling of that. Despite mixing as I go, I still have a definite mix phase, though when I have also recorded the song, I find it very difficult to mix it straight away. I need 10 days or so away from the record to be able to approach it fresh again, and be more objective about things and also not be emotionally attached to things in the mix.

"In the case of 'Take Me To Church' I would have started off where we left it after recording. The rough mix is all about getting the track to feel good, and I then get into the sonic details and the panning and all that kind of stuff. My first step with this song would have been trying to make the drums sound good, putting samples underneath the kick and the snare drums, and using other treatments."

Drums: Empirical Labs Fatso, Focusrite Red EQ, Drawmer DS201, Bomb Factory Sansamp PSA1 & Audio Ease Altiverb.

"Right at the top of the session is something called 'Asian Kit', which are samples from a preset in Logic that Andrew really liked. It features on quite a few of the songs on the record, mainly for handclaps and woodblocks and things like that. Further down is the Fatso track, with the M7/Fatso signal chain I mentioned above. I nowadays have a Mackie summing mixer on which I also have my Fatso and I will send a drum balance through that as parallel compression. In the past I would record the Fatso-compressed drums back in, but I got so bored of the latency issue and having to move stuff around that I prefer to have the Fatso running live most of the time.

"Below the Fatso track are the kick and snare tracks and samples, on which I used the Focusrite Red EQ, which I like a lot. The 'D' on one of the snare tracks is a Drawmer gate. Further down are the tom tracks, overheads, and an M7 and an M58 track, all, again, heavily edited. The M58 track also has the Drawmer, and the Sansamp for some distortion. Then there's a ride, an OH duplicate, and finally the 'Kit Plate' aux track, which has an Altiverb with the EMT Wendy Carlos plate reverb. That's the entire kit. There are not many sonic treatments on the drums. The Fatso sound is the main thing. If I was starting out today, it'd be the first thing I'd buy. I only use it on drums, but drums are such a crucial part of the recording process!"

Guitars & keyboards: TL Labs TL Aggro, Avid Trim, Waves Puigtech EQP1A, Eventide H3000, Focusrite Red EQ & Audio Ease Altiverb.

"Below the 'Kit Plate' aux track are four organ tracks. The top two were recorded through an amplifier via my 57 and Delta mics, and below that is a distorted organ track, with the Trillium Lane Labs TL Aggro compression plug-in on it and a Pultec EQ. The 'Organ 3000' track is probably me putting one of Andrew's Logic organs through my Eventide H3000. Further down are six guitar tracks without any treatments - the 'T' on one of them is a Trim - and below that are the piano tracks, some of which are again heavily edited. Andrew wasn't as good yet at playing the piano. He's very good now, though! I got him to do the left hand separately, which are the 'New Piano' tracks. I once again had the Focusrite Red EQ plug-in on the piano, and there's a 'Piano Verb' aux track, which also has the Wendy Carlos EMT plate from Altiverb, just with a different pre-delay. In general I like to create a sense of everything coming from the same space, so the Wendy Carlos plate is on many things in this song. There's more piano below that, again with just Focusrite EQ. In all there's a lot of piano, it was one of the most important elements of the song."

Vocals: Bomb Factory 1176, Focusrite Red EQ, SoundToys Echo Boy, Line 6 Echo Farm, Knas Ekdahl Moisturizer, Waves Renaissance Compressor, Avid Trim & Audio Ease Altiverb.

"The two 'tele' voices at the top are backing vocals. I think they were 'oohs' and 'aahs' that came from the demo. They don't sound like telephone voices at all, so I'm not sure why they're called that. Instead they almost sound like church voices. Below are two aux tracks for those vocals, with Bomb Factory compression, Focusrite EQ and an Echo Boy delay. The backing vocals below that - 'BV1-4' - are the ones with the signature Leslie effect on them, and they again have some Focusrite EQ and Bomb Factory compression, and I gave them some Echo Farm slap delay to add depth to the reverb.

"The next tracks are the verse lead vocals from his Logic session, called 'Male Creamy Lead Vocals', on which I put some Renaissance compression and a Trim, and 'Vox Verb' is the Moisturizer track, plus two Trim plug-ins, one being +1dB and the other -0.5dB. I hate automating in Pro Tools, so I will often use the Trim to bypass it. Writing automation is a tedious task. Below this are the chorus vocals, called 'New Vocal', so we obviously recorded these again at my studio. In order to make them sound like the demo vocal I stuck them through the same chain and EQ'd them heavily. 'Vox Plate' is the Altiverb, and there's a delay track below it which again has the Echo Farm, on a delay setting. Finally there are several more double-tracked chorus vocals, and they also have the Renaissance and Focusrite, and the Moisturizer and Echo Farm aux tracks below them."

Final mix: Neve 33609J, Manley Massive Passive & Klein+Hummel mastering EQ.

"The six master tracks are there to bring the levels down for certain things. Some things probably were too hot and the D-A converter couldn't handle it. At the bottom of the session are two mixdown tracks. My stereo mix goes through my Neve 33609J compressor, and then my Manley Massive Passive tube EQ, and finally my Klein+Hummel EQs. The K+Hs are the first solid-state versions, and they are incredible. I love them. I record my stereo mix back into the Pro Tools session via my Burl B2 Bomber. I normally work at 24-bit/88.2kHz, but this session was in 24/48 because that's what Andrew had worked with in Logic. One of the two stereo mix tracks has the Massey L2007 mastering limiter on it to give the artist and the record companies a better idea of what it will sound like after mastering. I send the mastering engineer both versions, so he knows what the artist has been listening to, but also has the freedom to do what he wants."

A key part of Rob Kirwan's role as producer with Hozier lay in identifying the elements that made his sound individual, and enhancing them. "To be blunt about it, Andrew is a white guy who sounds a bit like a black guy, and his music has a lot of black music influences. But there are many white soul singers out there today who produce really bland music. That was the danger. We could not go down that road. Andrew's voice and songs are exceptional, and we wanted signature sounds to match and enhance them. The guitar sound I just described [see main text] was a signature sound, as was a really tough drum sound, which is really important to me. The record had to sound dirty and tough, otherwise we would have strayed into that white soul singer territory. There was one song on the record that was quite poppy, and we spent a good deal of time trying to make it less sweet-sounding.

"Andrew's backing vocals, and the reverb on his lead and backing vocals, were two other signature sounds. The gospel and Irish church and folk music influences you can hear in Andrew's voice come from him. He was a member of the Trinity Orchestra, as a vocalist, and he also had his experiences with An na, of course. He is amazing at vocal harmonising. Most people will harmonise with a third or a fifth below and above, and that can sound really good, but Andrew would come up with these really strange but beautiful harmonies, that I thought sounded Irish, and he'd then sing them in a gospel way. I think this is a big part of the sound of the record that people are enjoying. It's a new sound, built on traditional elements.

"Another characteristic sonic element is the Leslie simulator plug-in on his Logic rig that Andrew had stuck on his backing vocals. The first time I heard it, I immediately knew that it would be an important signature sound. It was a real hook that we could hang the songs on."

"I'm a great believer in the artist's performance being the most important element of the song," insists Rob Kirwan. "It's one reason why I normally prefer for the vocalist to do his vocal performance once the entire track is recorded, so they can sing to that. It's also the reason why my studio is a lovely place to be in, with high ceilings and huge windows, and for a lot of the recordings we had the windows wide open. We're in the middle of the city centre, and there'll be loads of noises in the background, even seagulls, and I don't care. Of course, if there's the sound of a drill in the background during a vocal take, we'll do it again, but if there's the sound of a seagull flying by, so be it. All these natural noises, also of people moving and breathing while performing, are a part of somebody making music, and I like that, unless it really gets in the way. Conveying the sound of music being made by humans is what I'm here to do!"

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Inside Track: Hozier 'Take Me To Church'

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Producer and engineer Rob Kirwan oversaw Hozier's commercial breakthrough - by concentrating on the least commercial aspects of his music.

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Inspired by Norman Petty and Sam Phillips, studio owner Dean Amos has lovingly built a perfect recreation of a 1950s recording environment - right down to the dimensions of the pegboard!

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Vocal recording and production masterclass

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Vocal recording and production masterclass

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Everyone knows how much a classy vocal sound can add to any recording, but achieving this in the studio can seem something of a black art, so this month we demystify the recording, processing, and mixing techniques required to produce professional results.

In theory, getting a good vocal sound should now be easier than ever, as studio quality microphones seem now to be available for little more than pocket money. However, we still get lots of letters and emails at the SOS office about problems that musicians are encountering with this task, and we are often called on to advise on this topic in our regular Studio SOS visits. So I felt it would be valuable to put all the pertinent information in one place. Clearly the vocalist will always be the most important factor - a great vocal recording takes a great singer - but if we assume for the moment that the singer is delivering a good performance, what can you do at the engineering end to capture this in its best possible light?

Most pop vocal work is done using some kind of large-diaphragm, cardioid-pattern capacitor microphone, but there are exceptions. For example, some singers like to work with dynamic mics such as Shure SM58s, because the tonality suits their voice and/or the style of music. It's also perfectly possible to make a good vocal recording using a small-diaphragm capacitor microphone and, in the right room, an omnidirectional microphone can give good results, so it's really a case of the end justifying the means.

Perhaps more relevant than the type or make of microphone is its tonal character, as matching the microphone to the singer is more important than many people appreciate - and if you've never had the chance to compare a range of different mics in a recording situation, that's quite understandable. Even now I still get phone calls from people asking which mic is the 'best buy' within a certain price range, but that's really impossible to answer in a meaningful way without knowing something about the voice being recorded. Most studio microphones, especially large-diaphragm models, have a distinct tonal character, so it's best to have a selection of microphones available, allowing you to select the preferred one for each recording task, even if that choice is only between two or three different models.

If your budget can reach this far, then the obvious choice is to get one fairly neutral mic, one with a warm, syrupy quality, and another with a bit of sizzle on the top end. You may also wish to consider a tube microphone, though these can sound very different - so different, in fact, that it's hard to say what the tube sound is. To me, the better tube mics sound crisp and clear without being harsh or sibilant, and they also add weight and density to the lower vocal frequencies. Those mics that deliberately overdrive the tube to get a 'warm' sound can end up sounding mushy, dull or just plain distorted. Fortunately most mic manufacturers have now realised that the best-sounding tube mic is the one that doesn't use the tube to create artificial distortion - what it contributes naturally is quite enough.

Now I wouldn't dare suggest that women are in any way 'difficult' as a species, but I have found that choosing the best microphone to suit a female singer is generally more difficult than for male singers. Maybe it's because their voices tend, on balance, to be smoother sounding than men's voices, which probably equates to less harmonic complexity - or at least, a lower level of some of those complex harmonics. This leaves the throaty elements of the sound more exposed, so if you pick the wrong microphone you can end up with what sounds like distortion when what you're really hearing are the less desirable throaty artifacts of the sound being emphasised. Close miking is pretty unforgiving of unattractive vocal characteristics anyway, but if you're unfortunate in choosing a mic that exaggerates these less attractive artifacts, you're in trouble from the start.

As you might expect, microphones with a strong presence peak in the upper mid-range are the most likely to give rise to problems of this kind (presence peaks up above 8kHz are less likely to aggravate this type of problem), but then that same mic might rescue a session where you're dealing with a singer who has a voice that lacks definition. It's not usually a case of 'good mic versus bad mic', but rather 'suitable mic versus unsuitable mic'. Of the microphones I've reviewed over the past few years, very few of them have actually been poor from a technical standpoint, and even then, it's usually been from the perspective of noise rather than tonality.

The same tends also to be true for microphone preamplifiers. It's undeniably true that expensive boutique models can sound noticeably better than budget models, but I've used preamps and mixers costing under £100 in combination with budget Chinese capacitor mics and had great results, both artistically and technically. That's not to say that the very best gear wouldn't sound even better, but in many of the situations I've encountered, the equipment has not been the limiting factor. More often than not, it's what the microphone picks up that's the main problem with the recorded sound.

The usual choice is a large-diaphragm capacitor microphone as many of these are designed to have a specific tonal colour, often described as warm, airy, smooth, sizzly and so on. By picking a model to suit the vocalist, you can produce an optimal recording, but by the same token, the wrong mic can make a singer sound worse than they should. Small-diaphragm mics tend to have less off-axis coloration, and in general they are more accurate than large-diaphragm models. In my experience, they often make fine vocal microphones, although the large-diaphragm world does give you more choice over tonal colour. Nevertheless, large diaphragm mics now seem to have been adopted as the established, traditional method of recording vocals.

Most people know that microphones 'hear stuff', but they're not always aware of what stuff they hear. If you stick a singer a few inches away from the microphone, then obviously you're going to record the singer, but you're also going to record any other sound approaching the microphone. Vocals are almost always recorded using cardioid-pattern microphones, as they have a directional characteristic that excludes more off-axis sound (such as room reflections or spill from other sources) than omnidirectional microphones, which are equally sensitive all the way around. If you have a very dead room or vocal booth that will allow you to use an omni-pattern microphone, you may find that you get a more natural or 'open' sound than you do from a cardioid microphone, with the added bonus that there will be no tonal variation if the singer changes position slightly while singing. Omni mics also don't exhibit a bass boost when used close up (the so-called 'proximity effect'), so the amount of low end they produce doesn't vary as the singer gets closer to the microphone.

The other time an omni microphone might be preferable is when you're recording in a live environment where you want to capture the natural ambience of the room. An obvious example of this is choral music, but it can also be applied to solo parts where artistically suitable. Where you have a choice, a single-diaphragm, dedicated omni mic generally produces a more accurate off-axis response than you get from a multi-pattern mic that combines the outputs from two capsules to create the different polar patterns.

One further polar pattern to mention is figure of eight, because there's recently been a resurgence of interest in ribbon microphones, which all have a figure-of-eight pattern. Ribbon mics have a distinct tonality that is different to both normal condenser and dynamic mics, and they can sound good on a vocal that needs its high end smoothing out a little. These mics are equally sensitive to the front and to the rear of the capsule, so putting an absorbent screen behind the microphone while recording can help prevent unwanted room reflections from getting onto the recording. Figure-of-eight mics can be useful when recording a singer who plays acoustic guitar at the same time, as you can arrange for the dead zone of the mic (90 degrees off-axis) to be aimed towards the guitar to minimise spill. Note that all figure-of-eight mics have a strong proximity effect, so, again, if you work too close to them, the bass end will increase very noticeably.

Returning to the subject of cardioid microphones, one factor that hasn't attracted the attention it deserves is that single-diaphragm mics often behave differently to dual-diaphragm models. Dual diaphragms are used in switchable-pattern mics, and in cardioid mode they seem to pick up more of the room sound in close-miking situations than single-diaphragm models. The single-diaphragm mics always seem to produce a drier, more focused sound, which helps in situations where the amount of acoustic treatment that can be applied is limited. On the other hand, the dual-diaphragm model may deliver a more open sound, but often needs to be used in a more controlled environment to be effective.

Once you graduate from recording in your bedroom and start thinking about converting your loft or garage into a studio, the idea of a separate vocal booth can seem very appealing. Done properly, a vocal booth is a great asset, but simply creating a cupboard-sized enclosure with a door and then lining it with a couple of inches of acoustic foam invariably leads to disappointing results - what you end up with is a sound that is dull and lifeless at the high end, but uncontrolled and boxy at the low end. The reason this happens is that any absorbent material such as foam needs to be significantly thicker to be effective at low frequencies than it does at high frequencies. Where the foam is only a couple of inches thick, its effectiveness is much reduced below 400Hz or so, leaving those lower frequencies to behave almost as though the room hadn't been treated at all. What's worse is that by drying up all the high-end and upper mid-range reflections, the tonal balance seems worse than it was before, as the boomy low end now overwhelms the top end.

To create an effective vocal booth, the enclosure needs to be large enough to allow you to lose at least six inches from each of the three non-door walls, which will allow you to use a suitable depth of trapping. Four-inch foam or Rockwool slab on two-inch spacers to keep it off the wall is reasonably effective, but you should also consider leaving some small areas untrapped to add a little high-end reflectivity back into the booth, otherwise it may still end up sounding too dull. Another alternative is to fix small areas of reflective material to the surface of your traps - we used old CDs during one Studio SOS visit and it worked really well. Other approaches include putting the trapping behind perforated metal, plastic, or MDF sheeting.

As all microphone pickup patterns are three-dimensional, 'the sides' also pick up from above and below. Even if your room is quiet and there's no traffic noise, unless the room has been designed by a studio acoustician there will be reflections from walls and other surfaces that bounce 'second-hand' vocals back into the microphone. These reflections will be captured in addition to the direct sound of the voice, so you'll get the sound of the room overlaid on the sound you're actually after.

This is often a significant problem in the home studio, as there are reflective walls, ceilings and furniture conspiring to corrupt your vocal sound. While you might add reverb afterwards, you'll find that it doesn't hide the boxy tone caused by room reflections. Even people who've bought seriously professional-grade equipment fall foul of this problem. There's a parallel here with photography: even a budget camera can give you an artistically fantastic picture if what is in front of the lens looks good and is correctly lit. But if the subject is badly lit, then no amount of upgrading will bring about a significant improvement in the end result.

Regular readers will know that there is a way to improve this situation without resorting acoustic treatment of the whole room. All you need to do is improve the acoustics in the vicinity of the microphone. I think most people appreciate this on an instinctive level and, because I keep banging on about it, some musicians have tried using duvets for acoustic treatment. However, a surprising number simply hang a duvet on the wall and then sing facing the duvet! This may help a little, as the duvet will absorb some of the sound that would otherwise bounce back off the wall, but it's far from the best solution. Think back to the cardioid mic's pickup pattern and you'll see that in the above scenario its least sensitive area (directly behind it) is now facing the duvet, while its sensitive front and sides are still picking up room reflections.

A better option is to hang the duvet behind the singer, as any reflections coming past the singer's head will otherwise be picked up by the extremely sensitive front of the microphone. To reduce side pickup, you can hang a double-size duvet in a curve or use two to form a 'V' shape and have the singer stand with their back facing into the apex. That still leaves the ceiling and floor as possible reflection sources, but a rug or carpet on the floor will help enormously. If the ceiling is low, a foam panel above the mic and singer will do the trick nicely.

Improvised acoustic treatments made from duvets or blankets are effective at high frequencies and in the mid-range, but their efficiency falls off at low frequencies. In most cases, they work adequately over the vocal range, but doubling up the thickness or using a heavy, winter-grade duvet will improve their performance at the low end - useful when recording baritones. Such 'blanket' absorbers also work best when spaced away from a wall, so the best option, where practical, is to set up the vocal mic and absorbers out in the room, well away from the walls, but at the same time avoiding the very centre of the room, as that attracts its own set of special problems associated with room modes based on the room dimensions. As a final fix, you can shield the rear of the mic (after all, even a cardioid mic isn't totally deaf in the rear) by fixing a piece of acoustic foam close behind the microphone. Auralex make some neat devices of this type in triangle and butterfly shapes that slot onto the mic stand, but if looks don't concern you, a chunk of furniture foam will serve almost as well.

Even great vocalists sing better on some takes than on others, so it is common practice to record the same vocal part several times and then compile (or 'comp') the best phrases together into the final take. This is easy to do in a typical DAW as you can view the various takes on adjacent tracks, audition them, then separate out the phrases you wish to keep and move them to a new track. If you only cut in between phrases, your edits shouldn't be audible, but if you do have to edit during a continuous phrase, try to edit either just before a hard consonant or during a sustained sound such as 'eeh or 'aah', using a short crossfade to hide the transition.

Breath noise is a natural part of singing, and if you remove it completely, the resulting track can sound unnatural. You can reduce excessive breath noise by manually selecting the offending sections of the audio file and doing a destructive gain change to reduce the level by a few decibels or, if you're a little more cautious, you can use mix automation to do the same thing. It's also often possible to use a gate processor, setting the amount of gate attenuation when closed to around 6dB, rather than having it mute everything completely. The same techniques can be used to reduce headphone spill in sections where there is no singing, but this problem is easily avoided by using closed headphones in the first place. If your singer likes to work with one phone off, make up a special lead so that only one of the phones is working, as shown in the diagram above and explained in more detail in the article on cue monitoring in SOS January 2003.

Having got this far, can you now get on with your recording? Almost, but first you need to fit a pop shield a couple of inches in front of the microphone, otherwise you'll almost certainly get popping and banging on plosive sounds (usually 'B' and 'P' consonants), regardless of whether you're using a cardioid or omni mic. This is simply due to gusts of air being expelled from the mouth when a singer creates these sounds and those gusts slam into the mic diaphragm, producing a loud thump. A fine-mesh shield between the mic and singer will break up those gusts and fix the problem.

Various commercial nylon and metal mesh pop shields are available, but you can improvise your own from nylon stocking material stretched over a suitable hoop, if you prefer. Note that the foam wind shields that come with some mics are not very effective as pop shields, and they also tend to compromise the tone by filtering out some of the high end. I've covered this point many times before, but I make no excuse for repeating it, because I still come across people trying to use microphones with no pop shield. Even if you're using a stage dynamic vocal mic for recording vocals, a pop shield is still necessary to guarantee freedom from popping problems.

While a microphone shockmount is not mandatory, it really helps cut out low-frequency vibrations coming up from the floor, as they are prone to do if the singer has a habit of tapping their feet! This can be a real problem on wooden floors, so a shockmount is highly recommended. Some mics come with a shockmount, but if not, inexpensive generic mounts are available for most microphones - they not only solve vibration problems, but also look cool.

Now you're ready to record, and given that modern recording systems offer so much headroom, I'd suggest not using compression or EQ at the recording stage, as recording flat leaves you more flexibility for adjusting the sound afterwards. More experienced engineers sometimes process as they record, but as you may have already discovered, what sounds right in isolation may not always sound right once all the other parts have been recorded, so until you have gained the necessary experience, it's safer to leave your options open. Just make sure you leave a few extra decibels of safety margin on your recording system, as it's not only guitarists who get louder when the red light goes on!

You'll need to set up a suitably balanced headphone mix for the singer, and in most cases giving the singer a little reverb in the phones will help their pitching and also make them feel more confident about their performance. Make sure you don't record the reverb, though, as you'll want the freedom to adjust it at the mixing stage. Your 'bedside manner' can also make a big difference to the quality of the performance so try to be encouraging rather than critical, but at the same time make sure you capture enough takes to make up one good one.

If you intend to use a pitch-correction program or device, such as Auto-Tune, always apply this after recording - if the corrected sound is fed into the singer's headphone monitor mix, it will completely undermine their ability to control their pitching, and will result in a disastrous performance. Subdued lighting can also help get the singer in the right mood for the performance, but make sure you also look after the practicalities, such as making sure there is cool drinking water on hand and that the room is a comfortable temperature. Once you have captured that perfect vocal performance, you can polish it using EQ, compression, and reverb at the mixing stage, and I have a number of suggestions for tackling this in the most effective way.

No recording textbook would be complete without a discussion on de-essers, but in my experience they are very rarely needed. A de-esser is a technological solution to the physical problem where some singers produce very pronounced 'S' and 'T' sounds because of the way air moves around their teeth. In most instances, using a different microphone or moving the mic slightly above or below the singer's mouth is enough to reduce the problem to manageable proportions. You can also try fixing a pencil, vertically, right in front of the mic grille, holding it in place with a couple of elastic bands around the mic body. This shields the centre of the diaphragm, and definitely reduces sibilance, though the exact reasons have probably never been properly researched. Yet another solution is to put an extra layer of stocking material over your pop filter to attenuate the high end slightly.

However, where the problem is serious, electronic de-essing is sometimes necessary. A de-esser is, essentially, a special type of compressor that only reacts to frequencies in the sibilance range (typically 3-6kHz). Simple de-essers will reduce the level of the entire audio signal when the level of the sibilant frequencies rises above a threshold you set. Unless the degree of processing is very light, this can impart a lisping quality to the vocals. Models that attenuate only the upper half of the audio spectrum are better, but the very best ones attenuate only frequencies in the sibilance band, leaving other frequencies unchanged.

Once you have compiled the best vocal take possible and cleaned up any unwanted noise in the pauses, what next? If pitching is still a problem, you can use Antares Auto-Tune or one of its competitors to automatically force the vocal pitch to the nearest note in a scale you set, or to the nearest semitone. That said, in my experience chromatic correction gives problems if the singer uses a lot of vibrato or if they scoop up to notes. I find it's best to take the time to get the scale right, then use the slowest pitch-correction rate that will do the job properly, as this gives the most natural sound. You can always automate the rate of correction if some parts of the song need more help than others. Where more drastic correction is needed, you may be better off using a program such as Celemony Melodyne that allows you to fine-tune individual notes in a graphic environment. Less serious pitching issues can sometimes be resolved by processing individual words using a pitch-shift algorithm to push them up or down by the necessary few cents. You can even automate the amount of shift applied to different words or phrases if you have the patience.

When it comes to EQ, there are no standard settings. In an ideal world you'd have teamed the singer with the mic that best suited their voice, but in less-than-ideal conditions, removing some low mid-range in the 150-400Hz range can help clean up boxiness, while a broad boost at 8kHz and upwards can add 'air' and sizzle to the sound, but use as little as you can get away with. Using a low-cut filter to get rid of subsonic audio and breath rumble can also be useful, though this is best done at source if your mic or preamp has a low-cut filter.

Reverb is also pretty much a question of taste, but for most classic vocal sounds I usually end up going back to a plate emulation, as these don't try to impose any particular room type on the sound. Combining this with a suitable short-ambience treatment (there are some lovely convolution reverbs available) also suits most voices. By balancing the two reverbs, you can get anything from a fairly upfront and focused vocal to a long ballad reverb, and I've had some success making the attack of the plate reverb slower or adding some pre-delay, as this keeps it out of the way of the ambience reverb. Vocal plate reverbs rarely need to be longer than 1.8 to 2.5 seconds, and in many cases rolling off some low end below 200Hz gives a cleaner and more lively sound. Putting a subtle repeating delay before the plate reverb can also produce a nice effect, provided that the music has enough space to allow the effect to breath.

Should you wish to double-track the vocal part for any reason, the best results are obtained by doing it for real, getting the singer to duplicate their part on another track. You may even have enough spare parts left over from compiling your vocal that you can build a second track. Where the singer can't match the first performance closely enough, you have the option of slicing up the phrases and aligning them visually using your DAW's waveform display. Where this fails, you may have to resort to cheating using processing, though in my view none of the fake methods sound as good as the real thing. In the early days of recording, 'fake' double-tracking or Automatic Double-tracking (ADT) was often attempted using a short tape delay using a spare tape recorder. Somebody then tried to vary the delay time by slowing the tape reels by hand and accidentally invented flanging!

Although there are plug-ins that do a passable job of faking double-tracking from a single vocal, introducing pitch and time variation, you can get almost as good results by copying the vocal to a new track, then processing it with Auto-Tune or something similar, as this will create subtle pitch differences between the original and the copy. Adding a delay of 60-100ms to the second part can also enhance the illusion that there are two different parts.

Keeping the vocal level nice and even is generally done with the aid of a compressor, aiming for around 6-8dB of gain reduction (using a ratio of between 2:1 and 6:1) on the loudest points in the track. We've written a lot on this subject, but the optimal attack and release time settings depend on the type of compressor or plug-in you are using. If you're unsure as to how to set these, take a peek at some of the presets and see what values they used, but don't expect a preset to work without further adjustment, as the correct threshold setting always depends on the level and dynamics of the signal being processed. However, compression alone isn't always enough to tame the dynamics of some singers, so use your mix automation to bring up any words or syllables that are getting lost or to drop the level of those that stand out too much.

If you find it hard to judge how loud the vocals should be, try listening to the track from outside the door of your studio, which will make your mix sound more like it would on a domestic radio or hi-fi. From this perspective, the vocal should be clearly audible above the backing, but not so loud that it doesn't sound like part of the same performance. If in doubt, play a commercial record and listen to that from the same position. Ultimately this is a subjective decision and it takes experience to get it dead right every time.

If you're interested in learning even more about the topics covered in this article, then head over to the Sound On Sound web site, where you can find a huge number of related articles from previous issues of the magazine:

For more detailed information on choosing and using studio microphones check out 'What Mic Should I Buy First?' (SOS July 2003) and 'Getting Started With Condenser Mics' (SOS March 2005) for some pointers.

Managing any overdubbing session can be difficult, and vocal sessions can be particularly challenging. However, there's help at hand in 'Basic Overdubbing' (SOS March 2001), and you can also find advice on the important task of setting up the best headphone mix for the talent in 'Cue Monitoring Techniques' (SOS January 2003).

Equalisation is probably the most common processing tool applied to vocal recordings, so if you want a detailed explanation of what equalisers are, how they work, and how to get the best out of them, try 'Equalisers Explained' and 'Using Equalisation' from SOS July and August 2001 respectively.

Compressors are not only extremely common in recording studios, but they are also vital to almost all modern vocal sounds. For an in-depth look at the different types of compression, as well as tips on making the most of your own outboard and plug-ins, have a look at our two-part 'Advanced Compression: Part 1' masterclass that appeared in SOS December 2000 and Advanced Compression: Part 2 in January 2001.

We get asked questions about vocal reverb processing here in the Sound On Sound offices. Our 'Advanced Reverberation: Part 1' and 'Advanced Reverberation: Part 2' articles discuss all the common reverb types and algorithm parameters you might encounter, as well as going into some of the finer points of how to find the best settings for your particular track.

Mike Senior

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Vocal Recording & Production Masterclass

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Everyone knows how much a classy vocal sound can add to any recording, but achieving this in the studio can seem something of a black art, so this month we demystify the recording, processing, and mixing techniques required to produce professional results.

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Re-amping Guitars w. Eddie Kramer & Llexi Leon

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Llexi Leon and Eddie Kramer describe the re-amping technique used to record many of the guitar tracks during the Eternal Descent recording sessions in LAFX Studios, Hollywood. Filmed and edited by Andy McCreeth.

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General Bass guitar recording techniques

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General Bass guitar recording techniques

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FREE] Audio Masterclass with Joe Marlett (Credits: Foo Fighters, Blink 182, Fenix TX, Queens of the Stone Age) CLICK HERE TO WATCH NOW: https://prostudiolive.com/blog/audio-masterclass-joe-marlett/ Pro Studio Live and Signature Sound Studio bring you some of the most talented audio engineers to showcase their recording, mixing, mastering and production tips and techniques to the home studio and professional community.

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Steve Albini talks about some of his recording techniques.mp4

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The Fleacast guys visited with Steve Albini at his studio Electrical Audio on January 31st of 2011 to conduct an interview. In this segment Albini talks about working with different bands, their expectations, and some of his recording techniques. Search for The Fleacast on iTunes or visit www.thefleacast.podbean.com to hear the full interview.

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The James Jamerson Motown Bass Sound | Reverb Bass Tricks

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It is hard to speak of the bass guitar without mentioning the Motown recordings of the 1960's and 70's and their most famous session bassist, James Jamerson. Jamerson played on dozens of number 1 hits and changed the way bass was applied to pop music forever.

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For more great videos, subscribe to our channel! http://www.youtube.com/subscription_center?add_user=fullsailuniversity Full Sail University Advanced Session Recording Course Director Darren Schneider takes us inside the campus's flagship studio with singer/songwriter Kaleigh Baker to demonstrate the basic techniques to recording vocals. Check out more of Kaleigh's work here: http://www.kaleighbaker.com/ Note: Kaleigh's song "Sugar Daddy" was not recorded at Full Sail University.

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Full Sail University's Advanced Session Recording Course Director Darren Schneider goes over the basics of miking a drum kit. Microphones Used: Kick: Sennheiser e 602 Snare: Shure SM57 Toms: Beyerdynamic M 88 TG Overheads: AKG C 414 Hi Hat: Shure SM81 Learn more at http://w.fullsail.edu/fsmusicschool

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How To: Recording Acoustic Guitar Basics

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Full Sail University Recording Arts Course Director Darren Schneider provides basic tips for recording acoustic guitar that you can use in any recording environment, whether you're at home or in a studio like Full Sail's on-campus Audio Temple. Check out more music from guitarist F.C.

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Logic Pro X's Flex Pitch is an excellent implementation of what's now become a familiar and ubiquitous editing tool in modern production. It can be very simple to use-quick fixes can be made right in the Arrange window (Tracks area), by dragging pitch bars. Zoom in further, or switch to the Track Edit pane, and the more comprehensive Flex Pitch editing display becomes available, and in the Track Editor an automatic Pitch Correction slider can be pressed into service to do the job on an entire selection, region, or track, with the option of constraining that correction to a particular scale. And for more specific tweaks, just as with other pitch processing software, individual detected notes can be edited in pitch and time manually, by dragging on their bars.

And that's as far as a lot of people I know-especially less-techie musicians-go with Flex Pitch. But there are more advanced functions available in the main Flex Pitch display, including the 6 editing handles for each note-"hotspots"-and sometimes they provide the only way to deal with certain issues. So here are 5 tips for utilizing these additional Flex Pitch editing features. I'll focus on vocal editing, because that's probably the most common application for Flexing, but these suggestions could also apply to any instrumental track as well.

1-Use Flex Pitch Note Gain

By now, everyone is familiar with Logic's Region Gain function-you can use it to offset the level of individual Regions, prior to any inserts and channel controls. But if you needed to offset the level of just certain notes, say, in a melody-over and above any compression that might be applied to the track-then, traditionally, most people might turn to Automation. But Flex Pitch provides individual note Gain, via one of the editing handles (sorry, hotspots). Once the audio's been analyzed and the pitches detected, you can then get in and tweak individual note levels with ease-the waveform display even conveniently updates as you make the edits.

And of course, you can select groups of notes and drag on one of the handles to adjust them all at once.

2-Split up multi-pitch notes

Sometimes, a performer will slide into a new pitch so smoothly that the note detection algorithm doesn't pick up on the fact that it really should be a separate note. You might notice this in the display-a tell-tale pitch variation curve in part of the note suggests that what's really going on there is not some kind of extreme vibrato, but actually a different pitch. Fortunately, this is easy to remedy-just switch to the scissor tool, hover over the note bar in question, and snip. Flex pitch will break it up into two separate note bars, usually snapping the new note to its correct pitch line in the process.

Now you're free to individually edit and fine-tune the new divided notes as needed, with having to resort to fussing with the Vibrato and Drift handles.

3-Double-team pitchy notes with Drift & Vibrato

Speaking of the Vibrato and Drift handles, they can often be the secret weapon for taming a singer with what I'll call an "overly flexible sense of pitch". Sometimes you encounter a vocal part where the singer has a tendency to slide dramatically in and out of notes, and when holding a steady pitch-well, the pitch is anything but steady, instead embellished with a very intense vibrato. All this swooping and shaking may prove to be too much for the part, and while some auto-correction tools may try to address this kind of thing, it's usually better dealt with on a note-by-note basis.

This is where the Vibrato and Drift handles come in. If the singer has a very florid style, the pitch variations within a note that need to be tamed might include both sliding and wavering, so it'll take the combination of both Vibrato and Drift controls to tame them. Flex Pitch has three handles for this-Vibrato adjusts the pitch variation in the center of a note, while separate Pitch Drift handles at the beginning and end of each note address any sliding in or out of the pitch. I've found that, with overly dramatic performances, a little back and forth with these controls may be required, to get the notes in order-they tend to be interactive. So if I were to start by reducing vibrato, I'd probably have to then tweak both Drift values, and then re-tweak the Vibrato setting again, until I get the pitch variation line to be a little flatter.

But I wouldn't necessarily suggest taking the Drift and (especially) Vibrato all the way down to zero. At worst, that may make the note sound unnaturally even, and overly bland, eliminating too much of the human expression in the performance. Especially with lead vocals, I'd try to find a good compromise setting, one that preserves a little bit of the singer's characteristic style, but just keeps it in check enough to not be over the top. On background vocals, I might take a strong vibrato down a little further, but I still wouldn't try to eliminate it completely-we all know what happens when we're trying to perform transparent pitch editing and we take it too far (Auto-Tune, anyone?)

4-Roll your own harmonies

One of the creative things you can do with tools like Flex Pitch is creating vocal harmonies from a lead line. This can be just the ticket when the vocalist has already left for the day, and then the producer decides that some harmonies are needed, or for sessions where time and budget don't give the singer the chance to add additional parts after the lead is done.

There are two ways to go about it. You could utilize the automatic Pitch Correction slider, in conjunction with the Scale Quantize option, on a copy of the lead vocal track. Pitching the whole part up a musical third or fourth, while constraining the new harmony line to the key of the song, can sometimes produce a suitable, tight harmony, but a real, human-generated harmony is usually a little more complex. The harmony may not be added to every note in the main melody; it may vary between unison pitches and harmony pitches; often a good harmony line will move in opposition to the melody-climbing up as the melody drops down, etc. If I have to create a harmony from a copy of a vocal melody track, I prefer to do it by hand, in the Flex Pitch Track Editor window.

Audio Example 1: A vocal harmony line created with Flex Pitch: The original melody; the Flex'd harmony; both together:

It might seem like the long way around, but it's actually pretty quick, at least if you know what you want-dragging the pitch bars gives you a synthetic pitch reference tone, and you can speed things up by selecting notes in the harmony-to-be track phrase by phrase, dragging the entire phrase to the main harmony interval, and then just tweaking the notes that need to be altered, for the perfect harmony line.

5-Utilize the Formant tool

Lastly, there's one more handle that might not get as much use as the others, but can come in handy once in a while-Formant. Formants, as you might know, are the fixed resonances of the human voice-based on an individual's vocal cords and resonating cavities (head, chest, diaphragm), they're what give each voice it's unique, distinctive character. When something is pitch-shifted/transposed, the algorithm has to account for this, preserving the fixed formants as best as it can, or the formants will shift unnaturally, resulting in some degree of the famous "chipmunk effect". Editors like Flex Pitch do a pretty good job of this, but sometimes, if you're doing a more extreme transposition (more than 4 or 5 semitones), with some voices you may want to tweak the formant a little by hand, to restore a bit more of the natural tonal quality, just on those few widely transposed pitches.

Like I said, it's not something you'll probably be doing much, but every once in a blue moon, it can be helpful-especially when creating custom harmonies, like I just described.

So, Logic's Flex Pitch is more than just quick fix for pitchy singers-it can be called on to dig in and really fine-tune a performance, when necessary (and, ideally, only when necessary!), and also utilized for more creative application as well-and it sounds good, too. If you haven't really spent much time with it, it's worth a little extra effort to get to know the tweakier aspects of this powerful tool.
Learn more about Flex Pitch and Flex Time in this Logic Pro X video course in Ask.Audio Academy.
Joe is a musician, engineer, and producer in NYC. Over the years, as a small studio operator and freelance engineer, he's made recordings of all types from music & album production to v/o & post. He's also taught all aspects of recording and music technology at several NY audio schools, and has been writing articles for Recording magaz... Read More

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5 Tips For Working With Logic Pro X's Flex Pitch

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Logic Pro X's Flex Pitch is an excellent implementation of what's now become a familiar and ubiquitous editing tool in modern production. It can be very simple to use-quick fixes can be made right in the Arrange window (Tracks area), by dragging pitch bars.

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Free Downloads | Best Free VST Plugins, Free Sample Packs

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Check latest free VST, AU plugins for Windows and Mac. Effects, synths, drums, Kontakt libraries, best free sample packs, loops, presets, drum kits, sounds

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Hey what's going on out there? I thought I would share some ways to get "analog" style tape saturation effects using stock plugins in Logic Pro X. I've been getting a lot of questions about how to achieve this without third party plugins, so here you go! These are extremely useful tools that aren't so obvious to find. Enjoy!

Analog Tape Saturation Secrets (Logic Pro X)

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Analog Tape Saturation Secrets Using Stock Plugins (Logic Pro X)

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Hey what's going on out there? I thought I would share some ways to get "analog" style tape saturation effects using stock plugins in Logic Pro X. I've been getting a lot of questions about how to achieve this without third party plugins, so here you go!

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Learn when to choose different reverb types, whether to make your reverb dry or wet, how to set pre-delay and reverb time - and why reverb size matters!

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Logic Pro X's Space Designer lies at the heart of creating a mix with greater spatial depth and perspective. Mark Cousins unlocks its secrets

For this tutorial you will need the project file which you can download from here
When it comes to defining the 'spatiality' of your mix, Logic Pro X's Space Designer has to be the go-to plug-in for many engineers and musicians. In truth, though, many of us can take a slightly lazy approach to reverb - simply selecting a two-second Hall (or thereabouts) and routing through a group of instruments.

Listen to a commercial track, though, and you'll hear a well-crafted use of reverb: contrasting different room sizes, reverb colours and relative wet and dry ratios to create a coherent and existing acoustic space. Exploiting the full potential of Space Designer, therefore, is a tangible way to improve the quality of your mixes.

As the name suggests, the beauty of Space Designer is its ability to tailor the sound of the reverb. Although the wide selection of presets are a great starting point, the real benefit comes when you start tweaking various aspects of the reverb's sound.

An experienced Space Designer user understands how different sounds can work with reverb and how relatively simply controls - like pre-delay, or the tail's EQ setting - can be exploited to best effect. In this workshop, therefore, we look at the intricacies of Space Designer, from the basics to its more creative applications.

Reverb Nation

Logic Pro X includes a number of different reverb plug-ins, but it's Space Designer that's the real star of the show. While the other plug-ins create their reverb synthetically, Space Designer is powered by a technique called convolution.

The reverb in Space Designer is created using short Impulse Response (or IR) files that are acoustic snapshots of either a real physical space, or a vintage reverb unit. The process of convolution effectively combines the original dry source with the Impulse Response file, resulting in an output that sounds like it's been recorded complete with a room's acoustics.

To make Space Designer even more interesting, it also includes a 'Synthesized IR' option that lets you create your own Impulse Response from scratch. Don't be confused, though, as the Synthesized IR still uses the convolution process, and indeed, whenever you edit an aspect of the Synthesized IR, Logic will need to re-render the Impulse Response file that it uses for convolution. Although the Synthesized IR lacks the acoustic realism of the sampled version, it does have a few additional parameters that we'll explore later on.

Match Making

Matching the reverb to the source material usually starts by exploring the size of the virtual room, or to put it another way, the length of the reverb tail. Although there are no hard-and-fast rules, shorter, more percussive sounds tend to suit smaller reverb times, mainly to avoid the reverb masking the rhythmic qualities of the source material.

A drum kit, for example, tends to sound best in one of the Small Spaces settings, using a reverb time between 0.5 and 1.5 seconds. Sounds that are less percussive - like a string pad, for example - tend to work best with longer reverb settings, which also have the effect of placing the instrument towards the rear of the soundstage.

Rather than use multiple instances of Space Designer on a per-track basis, the best strategy is to create a range of three or four key sizes - starting from a small room, and ending up with a larger hall setting. Sending different parts of the mix at varying proportions will then allow to define the 'front to back' perspective in your track, letting some parts positively swim in a larger hall setting, while others simply gain a sense of a 'wraparound' acoustic space, without a noticeable reverb tail.

Colouring In

Beyond the basic size and duration of the reverb, there's plenty more to think about in terms of tailoring the sound to your mix. One key point that's overlooked is the relative colour of the reverb - contrasting between a darker setting that lets the reverb recede in the mix, and brighter settings that have a shimmering quality.

This tutorial is endorsed by Point Blank. With courses in London, online and now in LA, Point Blank is the Global Music School. You can study sound to picture on their Music Production Diploma courses, with pro industry tutors.

When it come to colour, Space Designer has two options. Firstly, you can choose to apply a static EQ setting using the EQ page, which changes the reverb colour irrespective of time. Secondly, use the Filter page to change the colour of the reverb over time, usually as a means of attenuating high frequencies over time.

Another interesting quality in the reverb is relative density, which distinguishes between the reverb heard as a series of discrete echoes, or a more cohesive tail. When you're working with Impulse Response files, the Density is an intrinsic part of the sample (although worth listening out for); but if you're working with the Synthesized IR option, you have the option to change density over time.

Plate reverb, for example, has a high density, which is why it sounds good on percussive sources, where as higher density settings can sound 'ringy' on non-percussive sources, where you may consider a lesser density setting.

Getting Creative with Space Designer

Although we've concentrated on reverb, the scope of Space Designer, and that of convolution in general, goes way beyond just the creation of an acoustic space. As the last section of the workshop illustrates, Space Designer can turn its hand to interesting and unusual sound-design treatments - ranging from drone-like resonators, to ghostly tempo-synced echo effects.

There are also a number of Impulse Response files taken from various speakers, so you can feed your sound through a variety of different cabinets and create some great lo-fi treatments.

Logic Pro X comes with a plethora of audio plug-ins, but it's the key players - Space Designer, the Compressor and Channel EQ - that are the cornerstones of effective mixing. Taking time to fully understand the intricacies of these plug-ins, therefore, will deliver the most tangible improvement in your mixes and the overall sound you achieve in Logic.

Focus On Complementary Plug-Ins

Although Space Designer works well on its own, it's worth noting a few additional plug-ins that pair well with it. The Direction Mixer, for example, is a great way of controlling the relative width of the reverb. Sometimes vocal reverb, for example, sounds better in a more monaural form, so that you can 'plant' the vocal and its reverb in the centre of the mix.

It's also worth playing with compression - either after the instance of Space Designer or, even more effectively, across a bus that both the dry signal and wet reverb are being routed to. The effect of the reverb pumping, as well as the extra intensity compression delivers, makes this pairing a musically fruitful partnership.

Step-by-Step - Reverb Basics

1:For the most flexible application, instantiate Space Designer across aux channels, fed from bus sends on the respective channel faders. Using the project file you can download at the start of this tutorial, create bus sends from the piano and drum tracks to two instances of Space Designer.

2: We'll start by looking at the reverb used across the piano track. Load a suitable starting point, either using the plug-in preset settings (organised by size), or you could simply load a pre-recorded IR file using the drop-down menu next to the IR Sample button.

3: As you've loaded Space Designer via an aux fader, the Dry level should default to a zero setting. To set the balance between wet and dry signal, therefore, either change the level being sent from the fader, or attenuate the aux return channel.

4: The IR Sample that you've chosen will have a predefined length, defined by the size of the room it was recorded in. To make the room artificially smaller, though, you can always reduce the Length parameter to a shorter setting.

5: The IR file defines the colour of the reverb. However, it's also possible to use the EQ tab as a means of adjusting the basic timbre. Try rolling off some low end, which can clutter the mix, as well as adding a touch of brightness.

6: Try experimenting with the Pre-Delay on the reverb. Increasing the Pre-Delay time can help create a greater distinction between the dry source and the reverb tail. The source feels closer, with the reverb floating ethereally in the background

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Logic Pro X Tutorial - Working With Reverb & Space Designer

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Logic Pro X's Space Designer lies at the heart of creating a mix with greater spatial depth and perspective. Mark Cousins unlocks its secrets For this tutorial you will need the project file which you can download from here When it comes to defining the 'spatiality' of your mix, Logic Pro X's Space Designer has to be the go-to plug-in for many engineers and musicians.

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Exponential Audio - Reverb & Effects Plug-ins

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Pro Tips For Using Reverb And Effects In Recording And Mixing

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Top engineers and prroducers share their tips for using reverb and effects when recording and mixing music and post.

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A muddy mix sounds bad. If there's mud then frequencies are fighting, nothing is popping, and it's difficult to hear each part.

It used to happen to me all the time. My mixes would blur together into a giant ball of meh. Everything sounded fine soloed, but my mixes lacked clarity and punch.

My tracks needed de-mudding.

It's time to start leaving the mud out of your mix for good. Don't waste time going back to clean it later.

Use these tips to get better at mixing music. Here's everything you need to know to keep the mud where it belongs.

It's an EQ thing

Fixing a muddy mix comes down to EQing.

EQing is adjustments you make to highs, mid-range, and lows of your sound.

Typically, a snare or a cymbal will register in the mid to high range. A kick drum or bass pad will show up in the lower mids or all the way in the lows.

The tricky part is that all sounds can register in the high, mid, and low frequencies.

For example, a snare or a vocal will tend to have some low frequencies that get cluttered up with all the other lows.

If you're not sure what frequency is all about Google's amazing new Spectrogram tool is a great way to visualize sounds. I recommend comparing the flute to the trombone.

Can't the lows all just get along?

Low end instruments also end up fighting to stand out on the same frequencies. This causes some major mudding.

The cluttering of frequencies is what causes a muddy mix.

Fixing these elements will make your mix clearer, crisper and punchier.

Follow these simple steps to de-mud all your mixes.

1. Cut the mud from the start

The easiest way to avoid a muddy mix is to ensure that you're working with the cleanest possible samples and recordings.

If your tracks are crisp and polished from the beginning there will be less unwanted noise floating around.

Plus, the cleaner your tracks are, the more responsive they are to certain processes. It will make the cleaning process much easier later on.

If you start with Grade-A sound, it's easier to get a Grade-A mix.

2. Panning against the Mud

When you're getting a first rough mix together panning is crucial to set you up for success later. Plus it's a good way to get a bit of the mud out of the mix early on.

Good panning will give each instrument its own space in the stereo image so it's not fighting other instruments.

If you have lead vocals start by leaving them in the centre. Same thing goes for bass. Once you have those centred you can pan everything else around them.

Panning is a creative process, so your pans are up to you and your ear.

But try to keep instruments that sound similar on opposite sides from one another. Don't hard pan all the way to the left or right unless it fits, just find a good medium.

You should also make sure that your mix isn't left or right heavy. Poor panning can make your mix lopsided. So check for balance often.

Hot Tip: use headphones and monitors to get a full sense of what your pans are doing. Using only headphones gives you an overly exaggerated sense of your stereo image.

3. Listen to every track soloed

You've probably already listened to your overall mix. That's how you know if it's muddy or not.

Now it's time to solo each track and pinpoint where the biggest bad boominess problems are.

Start by soloing your lowest tracks. These will typically be a bass drum, or bass guitar. It's best to start with your drums and go from there.

Listen for any unwanted boominess (don't try and fix it yet, just listen so you know what needs work).

Slowly unsolo each track and listen for which frequencies are fighting each other in that all-important mid-low frequency area.

To get a better sense of what's clashing, I like to use a multi-channel frequency analyzer like Voxnego's SPAN.

It's one of many helpful free VST plugins. It will help you visually see where frequencies are clashing.

This will give you a good idea of what needs to be fixed during corrective EQing.

4. Pardon me, but could you please pass the highs and the lows?

Your best friend for corrective EQing is the high and low pass filter. You'll notice a difference immediately.

When applied, a high pass filter allows only frequencies above a certain frequency to pass.

They're perfect for getting rid of unwanted low end on tracks that register mainly in the higher frequencies-like vocals or a lead synth. Which means less mud.

Most DAW software has a simple EQ for all your high and low pass needs. I used EQ Eight in Bitwig.

A low pass is filter is the opposite of the high pass. It only let's through the lows below the frequency you set.

Use it to roll off some of the highs that might be sneaking through into areas where you don't need them.

Feel free to use the high and low pass filters to free up your best possible sound. They're a perfect starting point for correcting EQ across your entire mix.

5. Carving EQ

The most common part of a mix that gets muddy is the 200-500Hz area.

Fixing it is as easy as carving out a bit of space in these frequencies.

Go back to your EQ insert on the tracks that are still sounding a bit muffled. Select the frequency range that you'd like to target and tweak it until it's sounding better.

Each audio track needs specific frequency settings and unique gain cuts.

So go through each track and carve out whatever sounds best for that track. But be careful. Removing too much can lead to a mix that's too thin.

In this step you may have to sacrifice some good frequencies on a track in order to benefit your whole mix.

Dig Deeper to Cut your mud

Like all audio production, It's important to start with the basics.

These tips will get you started with corrective EQ. But dig deeper into your mix and make changes on a micro level.

There are no overall best practices for EQing of this type.

Your music is unique. The only way to find your best sound is to tweak these concepts to your tracks.

That means listening to your mix on a deep level and applying corrective EQ that suits your track's specific needs.

Free your sound from that nasty mud. Your mix will sound punchier, clearer and overall better.

Which is what everyone wants at the end of the day: better sound.

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5 Tips To Avoid Ruining Your Mix With Muddy Sound | LANDR Blog

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A muddy mix sounds bad. If there's mud then frequencies are fighting, nothing is popping, and it's difficult to hear each part. It used to happen to me all the time. My mixes would blur together into a giant ball of meh. Everything sounded fine soloed, but my mixes lacked clarity and punch.

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How do you rescue a track when it seems like you've pushed it beyond repair? Never fear! Hollin Jones is here to explain how to bring it back to life...

1: Download an App
In many cases you will be attempting to repair audio that has already been recorded and has turned out to have some kind of issue with distortion, hiss or hum. In other cases such as when digitising vinyl or tape, you know in advance that the audio is likely to have some cracks, pops and other problems due to the nature of the source media.

When you record any source to digital audio it captures everything exactly as it sounds, unlike analogue recording media which can add their own flattering characteristics like warmth or compression during tracking.
SoundSoap will deal with distortion and cracks

2: Virtual Mic Plug-Ins
Room ambience can be one of the hardest things to get rid of on a recording but some plug-ins enable you to move 'virtual microphones' around in order to change the proximity of the source to the mics. This can take a while to get right but it can help you to 'reposition' a sound even after it has been recorded.

3: Don't Go Too Far...
When cleaning up audio it can be tempting to really throw everything at the problem. As when mixing, software lets us keep applying more and more processing to any sound without the likelihood of really running out of processing power. The risk with this is that you go too far.

In the case of audio repair this sometimes means layering on too much hiss removal which will blunt the high frequencies of the sound and thus deaden it overall. If a sound is too quiet you might also over-compress it in an attempt to make it louder, and correspondingly make it sound too squashed or forced.
Spectral repair tools such as Sony's Spectralayers enable you to get inside a sound

4: Specialist Tools
Regular DAW-style processing with mix tools such as EQ and compression can be useful but for more tricky jobs you will want a more specialised set of tools.

Spectral analysers and repair tools are the way to go here, with applications such as Izotope's RX and Sony's Spectralayers Pro leading the way when it comes to letting you get right inside a sound. Imagine you have an audio take with a vocal but halfway in there's the sound of a door slamming in the background.
Izotope's RX - another great spectral repair tool

5: Dedicated Tools
Some software has dedicated audio repair tools. It's often not DAWs but wave editors and audio-for-video packages that have them, as post production editors sometimes use these kinds of applications rather than DAWs since they're editing sound more than making music.

Packages such as Adobe's Audition and the sadly discontinued Soundtrack Pro for Mac as well as Sound Forge and Wavelab all have versions of audio repair tools.

Sometimes these are simply chains of plug-ins, other times they're able to analyse and repair audio clips with a single click. If you're repairing a large amount of audio this can be a more effective way to go about it than opening and searching through the waveforms of lots of clips. If you can't buy an app, try to borrow a system with it installed for a few hours.

6: Noise Print Apps
Some applications, particularly those designed for working with post-production film or TV sound, are able to analyse and take noise prints from digital audio clips.

The most common use for this is for identifying a specific sound, sometimes ambient, and then either creating a new clip with it or removing it from an existing file.

You can use it, for example, to 'match' ambiences by taking a noise print of a room with no other sound audible, and then applying it to a voiceover to make it sound like it was recorded in that room.

Or print from a few seconds of road noise then use that print to remove the same noise from a clip that you want to be quieter. It's a bit of a fine art getting it exactly right but it can be far more effective than using EQ in some cases.

7: Snapping & Stretching
When using material cut or copied from one part of an audio file to fill in or replace damaged signal in another your DAW or wave editor can really help you with tools like snapping or time and pitch stretching.

In some cases you may want to deactivate snapping in order to be able to stretch and position very precisely and avoid gaps or glitches.

8: Remove Spikes
Some of the most common errors in recorded digital audio are spikes and clicks caused by levels that peak too high during recording. If you open an audio clip in your DAW's sample editor you are usually able to zoom right in and get down to the sample level to locate these clipping events.

The more capable your DAW, the more likely it is to have this feature: and some wave editors have it too. Once you've identified the clip/ distortion event, which will probably look like a big peak in the waveform that flattens out against the top of the window, you can try to draw it out using the pen tool or select it and insert silence to remove it.
Getting down to sample level to identify spikes

9: Check Your Toolbox
Although dedicated tools can be very helpful, some audio repair tasks can be carried out successfully using the stuff that comes with your DAW or that you may already have as part of your music tracking and mixing toolbox.

Rumble and hiss, for example, can often be dealt with by using high or low pass filters respectively, and you can draw out or back off any other frequencies by using a regular multiband EQ.

Similarly, a multiband compressor can be used to tame or enhance specific frequency bands without adversely affecting other parts of the file.
Rumble and hiss can be sorted with a dedicated EQ

10: Re-Record
It might sound obvious, but if you have a recording where there's significant distortion, clipping or interference it's always preferable to re-record it if at all possible.

Thanks to broadband internet and cloud file storage it's now technically easier than it has been in the past to send someone a DAW project or just a reference file and get them to re-do a performance and send it back to you in high quality. Of course this is all dependent on them being willing and able to do it.

Sometimes the technicalities of a particular task make re-recording difficult or impossible to do. Nonetheless it should be your aim if it makes sense in the context of your situation. If you're lucky you might have access to a realtime online recording package like Steinberg's VST Connect which can make things easier.

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How do you rescue a track when it seems like you've pushed it beyond repair? Never fear! Hollin Jones is here to explain how to bring it back to life... 1: Download an App In many cases you will be attempting to repair audio that has already been recorded and has turned out to have some kind of issue with distortion, hiss or hum.

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Logic's powerful and intuitive set of equalisation tools can transform the timbre and spectral effectiveness of your mix. Mark Cousins is the equaliser...

Consider this - if you could produce a mix using only EQ or compression, but not both, which would you choose? I'd argue that while compression is beneficial to a good mix, equalisation is essential. Maybe two sounds are 'fighting' in the same part of sonic spectrum, or your DI'd bass needs extra kick around 80Hz - it's hard to imagine creating a mix without turning to EQ throughout the process.

Logic includes fully featured equalisers. More than just treble and bass controls, these can unlock a wealth of sound-shaping potential, from M/S equalisation through to matching two differing sound spectrums.

Equalisation in Logic comes in four principal forms - Channel EQ, Linear Phase EQ, Match EQ and Single Band EQ. The equaliser you'll use most of the time is the Channel EQ, with its two filters, two shelving controls and four bands of fully parametric EQ. One significant advantage of using the Channel EQ is the clear and informative FFT display, which gives an accurate visual representation of the sound spectrum you're trying to process. Tools like this can help you unlock your hearing's true potential.

Gain-Q couple strength

Like all good EQs, Logic's Channel EQ uses Gain-Q Coupling to ensure that there's some proportionality between the bandwidth and the amount of boost/cut. Imagine a small boost of 1dB with a relatively wide Q - as you raise the boost, the wide bandwidth becomes increasingly disproportionate, raising a large part of the entirety of the sound spectrum.

Gain-Q Coupling, though, creates a more focused bandwidth the more you cut or boost, maintaining the perceived bandwidth of the cut/boost. The result of Gain-Q Coupling is a more natural-sounding EQ. Note you can also adjust the Gain-Q Coupling - between Light, Medium and Strong, as well as three flavours of asymmetrical EQ - to suit your needs.

Looked at spectrally, you should start to see a few key points. Firstly, you should be able to spot the key fundamental frequencies - so, in the example included on the DVD, we can see strong fundamental frequency on the kick drum around 60-80Hz, the fundamental on the snare around 200Hz and a harmonic on top of that around 500Hz.

Above 1KHz, the sound spectrum gets increasingly complicated (with lots of extra harmonics) although these tend to be of a lesser amplitude than initial harmonics. This is where a sound's 'texture' can be heard; a big part of its timbral characteristics.

ZX spectrum

The key to using EQ is understanding where these frequencies sit in relation to the harmonic spectrum and how we can exploit them to our best advantage. Cutting or boosting around the fundamental or second harmonic can help change the instrument's presence in the mix, especially on low-end instrumentation like a kick drum or bass guitar. As you move up the sound spectrum, though, equalisation becomes more about broad 'colouristic' tones: so harshness and edginess might be tamed by a cut at 2kHz, while a boost about 10kHz can add a bright sheen.

For the six parametric bands, the parameters include the amount of cut or boost, the frequency, and the bandwidth or Q of the boost. The amount of boost tends to be subtler in the additive range (usually around 3-6dB or so), while cuts can be applied much harder (up to 10dB or more). The Q or bandwidth is also easy to see and adjust visually, often contrasting between broader sweeps of colour; with a wide Q, and more surgical adjustment using a narrower Q setting.

Another key area worth bearing in mind is the frequency area beneath the fundamental frequency. This is an area that's easy to overlook, but this 'top-and-tailing' (especially in relation to the low end) can really help a mix remain uncluttered. A simple Low Cut - either using the lowest filtering band on the Channel EQ, or the Low Cut setting on the Single Band EQ plug-in - can really aid the clarity of your mix, arguably defining sounds that have no interest in the low-end of the mix - like an acoustic guitar, for example - against instruments that have a stake, such as the kick drum or bass.

Another interesting tool is the Match EQ plug-in, which is a way of imposing the spectral properties of one sound on another. Match EQ offers a unique insight into the differences between two signals, as well as an alternative means of applying an EQ. Using Match EQ, can be an interesting way of discovering the timbral differences between your mix and a commercial recording or for matching the tones of electric guitars.

For the most predictable results, it helps if there's a degree of similarity between two sounds, rather than them being completely unrelated. Once Logic has analysed the Source and Reference, it will generate a new Match EQ profile that can be applied using various degrees of strength. Often a 'hint' of the matched EQ setting might be all that's required. Understanding both how sounds work, and how equalisation can be used to modify the spectral balance, takes time and practice.

Logic's equalisers are perfect tools to develop your hearing and refine your understanding. Take time to fully exploit the visual-analysis tools, and let them guide your ear in respect to improving your hearing.

Equalisation Step-by-step guide

1. The simple yet useful subsonic Low Cut setting reduces unwanted low-end. If you're not intending to apply other EQ on the same channel, the Single Band EQ is a good CPU-light option.

2. Single Band EQ has a range of creative applications. The High Cut setting can be used as a form of low-pass filtering, with the Slope setting the strength of filtering and Q Factor working as resonance.

3. This setting is similar to Little Labs' Voice Of God processor, which uses a resonant high-pass filter to increase bass energy. Set the Frequency to around 40-60Hz: increase Q Factor for added weight.

4. The Channel EQ is Logic's main equaliser, with eight bands to play with. The Analyzer is also useful to guide your EQing, even if you're clear about the principal frequency areas and how they sound.

5. The Analyzer's resolution can be sharpened to identify individual harmonics within the spectral profile. Click on the left axis to shift up or down, making the harmonic spectrum easier to see.

6. One useful trick with the Analyzer is to reduce decay to 0dB/s, so you'll see the cumulative spectral display over time: which, again, makes harmonics and frequency peaks all the more noticeable.

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Logic's powerful and intuitive set of equalisation tools can transform the timbre and spectral effectiveness of your mix. Mark Cousins is the equaliser... Consider this - if you could produce a mix using only EQ or compression, but not both, which would you choose?

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Producing a mix with a great low-end isn't just about EQ - it involves all kinds of creative approaches. Mark Cousins unlocks Logic Pro X's secret weapons for creating a truly earthshaking bass end to your mix...

Perfecting the bass end of your mix can be one of the hardest parts of the mixing process, either resulting in a muddy, ill-defined low-end, or a mix that lacks the rock-solid foundation of a professional recording. Although it's easy to think of one part of the sound spectrum being much the same as the other, the low end of the mix is a special case requiring a wholly different approach. Equally, there's a lot more to shaping the bass end of your mix than EQ alone, with Logic Pro X including many different creative options that go above and beyond what equalisation can offer.

Low-end theory

Before we delve into the specifics, it's worth taking time to consider the essentials of how the low end works in a good mix. First off, visualise your mix as having distinct and separate 'layers' of bass, with each part residing in a unique part of the sound spectrum. For example, a bass guitar might inhabit the lows and low mids around 100-200Hz, while an 808 kick extends down into the subsonic regions beneath 60Hz. Avoid over-populating each frequency area, and be ruthless about other instruments cutting into this prime real estate.

The second key point relates to perception of the stereo image and bass. Whereas our ears are very sensitive to the location of high-frequency sounds, they tend to be lacking when it comes to placing bass. The upshot of this 'locational insensitivity' is that a strong, centrally focused bass end will have far more impact than a low end that is unnecessarily divergent between the left and right speakers. In short, mono bass is good!

Pure Bass

Applying these principles, we can then start using Logic in a more informed way. Your main plug-in will be the Channel EQ, but before you start raising any lows, first think about the judicious use of high-pass filtering to keep key instruments out of the low end. Instruments like vocals, acoustic guitar and piano can all exhibit surprising amounts of low-end information, especially
if they've been close mic'ed. Place cuts around 150-300Hz, therefore, to ensure their reach doesn't extend too deep into the mix.

Even on instruments like bass and kick, you might want to consider using a degree of high-pass filtering. The distinction to be made is which of the two instruments - kick drum or bass - is the lowest element of the mix. If the kick is deepest, carve out some space using a 50Hz cut on the bass, and vice versa if you're contrasting a punchier kick with a low subsonic synth bass. This cut will become effective as you start boosting the low end, as frequencies below the 'musical' range won't be lifted at the same time.

Why phase linear?

If you're using an Equalizer on a parallel processing channel, ensure you opt for the Phase Linear Equalizer rather than the standard Channel EQ plug-in. In a conventional insert application, the differences between the Channel EQ and Phase Linear EQ are minimal - one sounds much the same as the other, in other words.

However, because a parallel application layers both equalised and un-equalised versions of the same sound, the small phase differences introduced by a conventional Channel EQ became much more apparent. In effect, the Channel EQ exhibits a slightly 'phasey' sound in parallel applications, noticeable at the frequency points where EQ is applied; the Phase Linear EQ sounds transparent, only highlighting timbral shifts you've applied

Beyond EQ

These fundamental aspects covered, our walkthrough deals with bass enhancement that goes beyond what EQ alone can deliver. Though often effective, EQ occasionally can't deliver that elusively deep low end to a mix, either because the frequencies aren't there, or the instruments simply don't drive the low end of the mix in the right way. The two techniques we'll explore use either parallel processing, or the active creation of additional 'subsonic' material.

Parallel processing tends to be associated with compression and the dynamics of a track, but it can have an interesting impact on the low end. Classic 'New York' parallel compression often combines EQ and compression to enhance the sound.

However, by loading the EQ more towards the bass end, and by combining it with midrange cut, we can create a 'thick and pumpy' low end that can be blended with the unprocessed source. By combining Compression and EQ, therefore, we create a bass enhancement that is both colouristic and dynamic, which is often far more exciting to the ear than a simple low frequency lift.

As well as compression, explore using Distortion across the parallel channel. Distortion, of course, is the secret weapon powering plug-ins like Waves' MaxxBass, and has long been used by engineers as a means of giving extra body to the low end. Logic's Distortion plug-in includes a Tone control, and lowering it down to 100Hz achieved a great low-end focused distortion, rather than typically shrill guitar distortion.

Not all sounds come with enough low end to give the desired amount of air movement. In these examples, it's useful to have the option of synthesising your own subsonic frequencies from an audio-track source.

Logic Pro X has two ways of extracting MIDI data from an audio recording - either using the Drum Replacement feature or, as part of the Flex Pitch feature, the ability to create MIDI data from Flex Pitch analysis. Once MIDI data is extracted, you can then synthesise the subsonic frequencies. Arguably, the most flexible synth for this is the ES2. Once you've got the trigger or note data extracted, you need to replicate the envelope of the sound, so the subsonics have parity with the source material they're trying to double.

What's most important is having a clear idea of what you want to achieve and how the various instruments relate to another. With most of us having less-than-perfect monitoring, it's worth having a 'conceptual' angle on your low end, even if you can't quite hear it.

Step-by-step: Better Bass

1. Our first two examples use Parallel processing to enhance the bass end. Create a bus send from the drum submix, and place a Linear Phase EQ plug-in on the newly created Aux fader.

2. On the Linear Phase EQ, setup a bass boost (+9dB at 80Hz) combined with broad, midrange cut (around 1kHz) and remove high end above 14kHz. Don't worry if the settings sound too extreme...

3. After the EQ, add a Compressor. Use a hard Ratio (around 4:1) and adjust the Threshold for plenty of gain reduction. Set a fast Attack and slow Release (500ms) so the compressor 'pumps'.

4. Set correctly, you should have a compressed version of the drums with lots of low-end 'wobble'. This can now be blended into the mix by raising or lowing the Aux fader to enhance the low end.

5. Disable the EQ and Compressor. Place a Distortion plug-in in the same Aux fader. Set the Tone control low, so that the Distortion is only added to the low end of the kit; add Drive to make it rumble!

6. Low-end enhancement works best in the centre of the mix. For focus, use an instance of the Gain plug-in with the Mono button engaged. The Aux signal is now totally devoid of stereo information.

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Producing a mix with a great low-end isn't just about EQ - it involves all kinds of creative approaches. Mark Cousins unlocks Logic Pro X's secret weapons for creating a truly earthshaking bass end to your mix...

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Replacing or augmenting your existing drum recordings is a breeze in Logic Pro X, and can open a wealth of percussive opportunities. Mark Cousins doctors his rhythms...

In an ideal world, every drum recording would be perfect - each part of the kit tuned to perfection, a collection of world-class microphones expertly placed, and just enough 'room' to let the kit breathe. In reality, though, this sonic perfection can be hard to achieve; and even though the performance might be spot-on, there's some part of the kit's sound that doesn't quite work. Maybe the kick drum doesn't have the low-end punch to hold the track together, for example, or the snare sounds dull and lacks the crisp high-end that your music deserves.

Although few engineers will openly admit it, the dark art of drum replacement is something we all need to turn to from time to time. As we'll see, this doesn't necessarily mean replacing the whole kit, but instead opens up a wealth of creative and sonic potential with even the most basic drum recordings. When it comes to Logic Pro X, it's also pleasing to note that the tools for drum replacement are neatly embedded into the application's workflow, negating the need for any additional third-party plug-ins that are necessitated in less-well-equipped DAWs.

Get the groove

If you're reprogramming parts of the kit, you'll want to make use of Logic's Groove Template feature, rather than simply using a straight 16th quantize. To extract the groove, either use a combined MIDI region with multiple kit elements (like kick and snare, for example) or perform a rough-and-ready extraction of the overheads. Select the MIDI region and, from the Inspector's drop-down quantize menu, pick Make Groove Template. Once you have your Groove Template it can be applied to any MIDI region, simply by selecting the newly added Groove from the drop-down quantize menu. Once applied, the MIDI will lock to the transient hit points you've previously extracted, and therefore be in perfect time with the rest of the kit.

Kick off

The process of drum replacement begins with the raw drum recordings. Ideally, you need a relatively clean, isolated track for each part of the kit you intend to replace. As the driving force of the kit, most engineers focus the majority of their drum-replacement energies on the kick and the snare, which is especially easy - thanks to these parts of the kit often having their own microphones and a good degree of separation. Channels like the hi-hat and cymbals, however, are far more difficult, largely due to the high degree of spill between the various channels.

To make things easy, Logic's drum replacement is a track-based editing feature, found (logically enough) under the Track menu. When you initiate the command, there are a number of things that happen so that the drum replacement is as immediate and hassle-free as possible. First off, Logic will look through the audio regions on the track and create a new MIDI region based on the transient points of each drum hit. The relative amplitude of each hit will also be transferred to a comparable MIDI velocity, so that the replacement process captures both the timing and the dynamic properties.

Pay threshold

As well as the MIDI region, Logic also creates a new instance of the EXS24 sampler, loaded with individual kit-piece samples. Later on, you can choose to replace the EXS24 with an instrument of you own choice, but the included sounds give you plenty of choice, as well as sounding good in either replacement or augmentation applications.

One key aspect of the drum-replacement process is setting the Relative Threshold as part of the accompanying dialogue box. The important point to note is that the MIDI region isn't a fixed entity, and that in actuality, raising or lowering the Relative Threshold will result in notes being added or removed from the region. The trade-off, of course, is getting the perfect balance between the threshold being too high; and ghost notes or quieter hits not being triggered, or too low; and other parts of the kit 'bleeding through' and triggering the wrong sound. It's only when you press 'OK' that the MIDI region is finalised, although you can always edit the MIDI data at a later point.

Regional news

With the MIDI region in place, you then have two ways of further refining the process, either editing the data itself or exploring alternative sound sources. A degree of manual MIDI editing is inevitable - either to edit the occasional incorrect trigger point, or to adjust the note velocity. When it comes to dynamic adjustments, the Velocity Processor (as part of the suite of MIDI plug-ins) is a real godsend. Use the Velocity Processor like a compressor - with a hard ratio to tame notes with a high velocity, or a more graduated curve to create a more even dynamic across the performance.

The creative possibilities of drum replacement really come to the fore when you start to explore sound possibilities away from the EXS24. One popular technique is to double the kick drum using a sound with an electronically generated low-end - like an 808 kick, for example, or a Sine wave. In this application, you might want to select an appropriate Trigger Note rather than default GM drum mapping, arguably picking a note number that matches the song's key signature. Virtual Instruments like Logic's Retro Synth are a good choice for subsonic bass sounds.

The range of excellent virtual drumming software (in addition to Logic's own Drum Kit Designer) means that you can actually take the route of complete drum replacement and achieve some surprisingly realistic results. The issue, of course, is 'replacing' the overhead tracks that won't have the separation of the kick, snare and toms. If the pattern's simple enough, consider reprogramming the cymbals and hi-hat, or use the drum replacement function to extract all the transient points, which you can then go in and map or delete accordingly.

Doctor rhythm

Logic's drum replacement feature is a great way to enhance and expand your drum recordings, even if you're starting with recordings that you're relatively happy with. The ability to add transient-accurate subsonic bass, or augment an existing snare track with a TR-808 clap, can really aid the sonic scope of your drum recording. Equally, the opportunity to extract the DNA performance and completely remap it to a new sound set means that the recording process is only the beginning of what you can achieve in Logic Pro X.

Step-By-Step: Better drums in Logic

1. Open the project included on the DVD, or another multitrack drum recording of your own choice. Select the kick-drum track as the first part to work on and then choose the menu option Track > Replace or Double Drum Track.

2. Once you've initiated the command, Logic will create a new instrument alongside an accompanying MIDI region based on some initialised settings. Pressing Prelisten will play back the replacement alongside the original kick drum to audition it.

3. You'll want to select a sound that closely matches or complements your existing kick drum. Scroll down the list in the Library to select from a palette of 31 kick drums. You can expand your options later on, so go with what works for now.

4. The crucial parameter with drum replacement is the Threshold. Pay close attention to the differences between the audio region and the MIDI region, raising or lowering the threshold so that it captures all the kick drum's hit points.

5. Press OK once you're happy with the settings, and then move over to the snare-drum track. This time, ensure you select the Snare option from the drop-down Instrument selection. This will ensure both the correct sound assignment and MIDI note.

6. Again, you'll want to pay close attention to the threshold. Set the threshold low enough so that it captures the ghost notes, but isn't triggered by the kick-drum spill. Use the MIDI region as a guide to what notes you've captured correctly.

This tutorial is endorsed by Point Blank. With courses in London, online and now in LA, Point Blank is the Global Music School. You can study sound to picture on their Music Production Diploma courses, with pro industry tutors.
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Percussive Opportunities: Better drums in Logic Pro X

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Replacing or augmenting your existing drum recordings is a breeze in Logic Pro X, and can open a wealth of percussive opportunities. Mark Cousins doctors his rhythms... In an ideal world, every drum recording would be perfect - each part of the kit tuned to perfection, a collection of world-class microphones expertly placed, and just enough 'room' to let the kit breathe.

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Delve deeper and you'll find that the Logic Compressor plug-in has a long list of tricks up its sleeve. Mark Cousins digs in and gets squashed For this tutorial you will need the project file - download here

Chain Gang

We first took a look at the revised Logic Compressor a while back exploring the key processes involved in setting up and using the plug-in. However, given the inherent flexibility of Logic Pro X's Compressor (which is as good as any third-party alternative you could care to mention) combined with the fact the engineers use compressors in some fiendishly inventive ways, it's well worth a closer exploration of the plug-in's deeper features. We're going to look at three techniques: frequency-conscious compression, New York compression and EDM Side Chaining. Together, these show just what a versatile performer the Compressor plug-in is...
This tutorial is endorsed by Point Blank. With courses in London, online and now in LA, Point Blank is the Global Music School. You can study sound to picture on their Music Production Diploma courses, with pro industry tutors.

The Logic Compressor - Frequency-Conscious

If you've mastered the basic parameters of compression - ratio, threshold, attack and release - you'll soon start to appreciate that a compressor's response can vary between different types of input signal, as well as variables like the model of the compressor. True mastery of compression, therefore, is about fine-tuning this performance - whether it's picking the right model type, for example, to flatter the sound you're processing, or making the compressor more or less sensitive to characteristics of the input.

One key facet of a compressor's response is frequency, or more specifically, how different frequencies can push a compressor into applying gain reduction. This is often particularly evident on drum compression, where the kick drum dictates the movements of the compressor more than any other part of the kit.

The reason for this behaviour is that the kick is supplying a disproportionate amount of sound energy, largely because our ears are much less sensitive in that part of the frequency spectrum. While the effect is interesting, the resultant pumping sound might not be the result you want to achieve.

Focus On De-essing

De-essing is another form of frequency-conscious compression - in this case, making a compressor particularly sensitive to sibilant sounds in the human voice. You could use the dedicated DeEsser plug-in, of course, but there's a lot to be said for the simplicity and visual clarity of the Compressor for simple de-essing tasks.

To make the Compressor sensitive to sibilance, activate the Filter section, using the BP (band-pass) mode and tuning the frequency to where the sibilance is most prominent (usually around 3kHz). Now fine-tune the threshold and ratio so that the gain reduction is only triggered when the sibilance occurs, using ratio to define the amount of sibilance reduction you want
to achieve.

To have more control in relation to frequency, we make use of the compressor's Side Chain section. The concept to grasp here is that there's a distinction between the sound present at the input of the compressor, and the signal being used to 'drive' the compressor.

The Side Chain section of the Logic Compressor, therefore, looks not entirely dissimilar to an equaliser, but in this case, it's not applying equalisation that we'll hear in the output stage, but instead an equalised version of the Side Chain being used to feed the Logic Compressor.

In the previous example of a kick-heavy compression, the solution would be a high-pass filter (HP) set just above the frequency of the kick drum (around 100Hz). To fine-tine the Side Chain, use the Listen button, which will allow you to audition the Side Chain directly in the output of the plug-in, remembering to put filter back to On when you're finished. Once active, the kick drum's input on the Gain Reduction should be far less severe, resulting in a more natural, transparent-sounding compression. What we've created here, therefore, is frequency-conscious compression, where the Compressor's response has been tailored to be more or less sensitive to specific frequencies.

1: Open the first project included in the workshop folder. Across the drum loop, instantiate the Compressor plug-in and set up a moderate amount of compression using the Vintage VCA model - 5:1 Ratio, -30dB Threshold, medium Attack and Release.

2: Open the Side Chain section using the button on the top right-hand corner of the Compressor. In the Filter section, press Listen to hear the Side Chain input feeding the Compressor. By default, this is a low-pass filtered version of the input.

3:Reconfigure the Side Chain filter so that it removes the kick drum. Use the HP (high-pass) mode, and adjust the frequency so that most of the kick drum's energy is attenuated. Deactivate the Listen mode and move the filter to On.

4: Toggle the Side Chain filter between on and off to hear the compression with and without kick drum attenuation. With the Side Chain in place the result is arguably more musical, and closer to how our ears hear the drums.

5: As an alternative, try adjusting the Side Chain so that the amplitude of the Kick is increased. This will create a deliberately 'abused' sound that ties the Compressor's movement to the kick. Use the ParEQ mode and apply a gain increase at around 80Hz.

6: Now listen to the compression in action, which is now sensitive to the kick drum's dominant frequency. You might want to adjust the Release times so that the movements are empathetic to the sound and speed of the kit.

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Logic Pro X Tutorial - Advanced Use Of The Logic Compressor

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Delve deeper and you'll find that the Logic Compressor plug-in has a long list of tricks up its sleeve. Mark Cousins digs in and gets squashed For this tutorial you will need the project file - download here Chain Gang We first took a look at the revised Logic Compressor a while back exploring the key processes involved in setting up and using the plug-in.

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Taking a more creative approach to FX routing can bring new life to an otherwise tired collection of plug-ins. Mark Cousins explores this sonic route...

In the search for music perfection, most of us look for the 'killer plug-in' that makes everything sound good - whether it's some form of magical compressor, the ultimate reverb, or a boutique EQ. In truth, though, it's often where and how we use our plug-ins that transforms an ordinary mix into something truly exceptional.

Rather than simply inserting one plug-in after another, isn't it about time you thought more about the routing and ordering of your plug-ins? Indeed, look more inquisitively at the signal path in your mix - using inserts, busses and a combination of both parallel and serial processing - and you may find that your music becomes all the better for it.

Channel Tunnel
Most plug-ins tend to get placed into a channel's insert path using the ever-expanding series of slots just beneath the Input assignment. From an operational standpoint, this approach makes sense, especially in respect to tagging an instrument with its own combination of plug-ins.

What you see, therefore, is a simple serial signal path where one plug-in's output flows into the next plug-in's input. The effect of the processing is cumulative, with each plug-in adding to the overall effect achieved.

Even in this simple setup, it's worth remembering that the order of the plug-ins has a big effect on the result achieved. Place a reverb halfway down the insert path, for example, and every plug-in after will affect both the dry signal and the reverberated.

Likewise, EQ before compression (rather than after) can affect how a compressor behaves, especially when large amounts of boost are used. Ultimately, none of this wrong (indeed, it might well be the effect that you're after), but it's important to remember some alternative approaches.

The second most common way of using plug-ins is with a buss send combined with a separate auxiliary channel. Typically speaking, the use of buss sends/aux channels is primarily reserved for ambience effects like reverb or delay, with the main advantage being that any number of channels can be routed through to the effect, with an individual buss send pot controlling the respective level.

What we've described here is an example of a parallel routing, with the reverb or delay running in tandem with the main channel fader.

A key advantage of parallel processing is that we can create some further distinction between where and how signals get processed. In the reverb example, plug-ins on our channel insert path only affect the dry signal, whereas plug-ins on the reverb's aux fader only process the reverb tail. An equalizer after the reverb, therefore, can be used to colour the timbre of the reverb without having any effect on the main signal that's on the channel path.

Having understood the principle differences between serial and parallel processing, therefore, let's start to imagine the myriad of ways plug-ins can be applied.

Firstly, it isn't the case that aux channels are only for reverb and delay. Plug-ins like compression and distortion work particularly well in a parallel setup, using the return level (on the aux fader) to set a balance between processed and unprocessed signal.

You can also achieve some interesting results placing a filter (either low or high-pass) on the aux channel ahead of the compression or distortion, so that the 'mojo' enhancement is only applied to a specific frequency band.

If you're using parallel processing a lot, it's well worth noting the routing options both in respect of channels and aux faders. Rather than using a buss send, you can route one or more channels directly through to a buss using the channel's output assignment. What's particularly interesting, though, is that any number of aux channels can be set to the same buss assignment, letting you create multiple paths running at the same time.

In the walkthroughs, we'll see how we can use this feature to create a parallel guitar amplifier configuration - whereby a single input source is routed to two parallel aux channels, each with their own FX path and guitar amplifier.

In effect, the setup mimics the sound of splitting a guitar and sending it through two separate amps in the same room, each with a slightly different FX chain and amplifier settings.

This parallel configuration is impossible to create using Amp Designer and Pedalboard alone, and, thanks to the routing (rather than simply duplicating a pre-recorded audio track) it's possible to be played through 'live'.

Rather than thinking of plug-ins as 'cumulative' processing, therefore, start to think about how they might individual contribute to the overall sound, and how they might want to interact with one another.

For example, do you want to compress the sound of the reverb? Do you want the delay taps with or without reverb on them? Often these small but important decisions can have a big effect on the effectiveness of your mix and, more specifically, the distinction you create between instruments and sonic features in your soundstage.

Obviously, the walkthroughs here only begin to scratch the surface of a more creative approach to FX routing. Each plug-in in Logic brings new possibilities in both parallel and serial operation, so it's well worth contrasting the use of plug-ins applied across channel inserts with an approach that incorporates elements of parallel processing using the aux channels.

Most importantly, you may well find that some of your most neglected plug-ins gain a new lease of life used in a more creative way, often creating effects that you didn't think possible.
Focus On Decluttering Your Mixer
If you're working on a laptop it's increasingly easy for Logic Pro X's mixer to engulf your screen, blocking your view of the all-important arrangement window. One solution is to par down some of the additional graphic elements and make the mixer closer to what it was in Logic Pro 9.

On the whole, I prefer a mixer without Gain Reduction meters (especially if you're using lots of third party compressors), EQ thumbnails, MIDI effects and Track Icons. You can find all of these options under the Mixer's local menu View > Channel Strip Components, as well as the option View > Hide Legend that removes the function labels on the left-hand side of the mixer.

This tutorial is endorsed by Point Blank. With courses in London, online and now in LA, Point Blank is the Global Music School. You can study sound to picture on their Music Production Diploma courses, with pro industry tutors.
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Logic Tutorial: Creative FX Routing - MusicTech

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Taking a more creative approach to FX routing can bring new life to an otherwise tired collection of plug-ins. Mark Cousins explores this sonic route... In the search for music perfection, most of us look for the 'killer plug-in' that makes everything sound good - whether it's some form of magical compressor, the ultimate reverb, or a boutique EQ.

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With a graphic refresh and a host of new features, Logic's delay plug-ins have never looked or sounded so good. Mark Cousins experiences the joys of repetition...

It's great to see a range of plug-ins receiving a much-needed facelift as part of the continual development of the application. From the vintage-styled Compressor to a range of sample-accurate metering options, Logic's plug-ins have not only received a visual upgrade, but also some important functional improvements. Arguably the most welcome developments have been the sonic functionality in Logic's Delay plug-ins - including Stereo Delay, Tape Delay, and Sample Delay - adding improved routing options, distortion and so on.

Like Compression, EQ and Reverb, delay is a great way of adding interest to vocals, synths, guitars and a range of other instruments. Recognising this, Logic Pro X includes three principal delays effects - the aforementioned Stereo Delay, Tape Delay, and Sample Delay - which can be used for a number of different audio effects. Each delay plug-in brings a unique set of parameters, as well as a definable sound, so it's well worth understanding how best to exploit these traits.

Tape Head

Whether or not you like 'lo-fi' treatments, I'd strongly recommend Logic's Tape Delay plug-in as your primary plug-in for delay-based tasks in mixing. Based on a single delay line, Tape Delay is arguably the most straightforward delay plug-in to work with, whereas the other two delay plug-ins - Stereo Delay and Sample Delay - have a separate delay tap for each side of the stereo signal.

As we'll see later on, despite the apparent appeal of a true stereo delay, it can also create its fair share of sonic problems that can impair the effectiveness of your mix.

The real beauty of the tape delay is, of course, its 'coloured' sound. Put simply, a delay sounds more musical when it's not a flawless repeat of the input - whether it's something as simple as a contrasting timbre or, in extreme cases, a touch of added distortion.

Compared to the Tape Delay plug-in in Logic Pro 9, which largely kept its signal degradation under wraps, it's pleasing to see direct access to the 'Character' of the tape delay - moving between a subtle rolling off of the high-end, for example, to the more saturated delay effects used in dub music.

Another unique factor in the Tape Delay plug-in is the Modulation section. This Modulation section can be thought of as adding warmth to the output, effectively though the introduction of pitch-modulation artefacts.

In the case of the Flutter controls, we have a direct emulation of the small pitch instabilities that occurred on tape (especially when the mechanism wasn't kept particularly clean), which you can add in varying amounts. The LFO control modulates the delay time, effectively creating the shimmering effect we know of as Chorus. As you'd expect, both of the Flutter and LFO controls come in really useful on guitars and vocals, where the extra warmth and pitch modulations sound pleasing to the ear.

Super Stereo

As the name suggests, Stereo Delay is specifically designed as a means of creating delay effects that vary between the left and right-hand channels. A classic 'ping-pong' delay effect, for example, will appear to jump from one side of the mix to other, usually working with a tempo-based time division. On its own, Stereo Delay certainly sounds impressive, but it can dominate the extremities of the sound stage.

In theory, the prime 'real estate' of any mix is either dead centre, or the extremes of left and right, and as such, you should avoid placing too many sounds in these all-important areas.

On the whole, the parameter set for the Stereo Delay plug-in is relatively straightforward, with an individual delay time for each channel, a basic low- and high-cut control, and two forms of feedback.

One prominent improvement in the way the plug-in works is the Routing control, which has a number of drop-down options that govern how the signal moves between the left and right-hand channels.

In effect, the Routing control works with the Feedback, Crossfeed and Phase controls, which would have previously needed to be set manually in order to configure effects like ping-pong delay. By quickly switching routing settings, therefore, you can explore a number of different styles of stereo delay.

In Sync

One feature common to both Tape Delay and Stereo Delay is the ability to lock delay time to a tempo-based division using the Tempo Sync button. In many cases, this is often the preferred application for delay, as exemplified by The Edge's tempo-synced guitar delay, where the effect becomes a rhythmic feature in its own right. In this respect, it's well worth exploring the syncopated delay settings (like 1/8 dotted), where the delays don't fall onto main beats.

While most delays nowadays tend to be tempo-synced, there's still a lot to be said for delay settings that don't fall on grid divisions. The classic Tape Slap effect, for example, uses a delay time between 50 and 100ms as ameans of doubling an existing sound, like a vocal or a guitar line.

Add a small amount of LFO modulation on the delay time, and you'll create what is often referred to as Automatic Double Tracking - or ADT - which approximates the sound of an instrument being double tracked. Again, ADT is something that has fallen out of use (mainly because infinite track counts make real double tracking easy), but although artificial, it has a sound of its own.

Delayed Reaction

Despite some strong third-party offerings, Logic's Delay plug-ins are assured, dependable solutions that still get plenty of daily use in our studio. The combination of low CPU load, plenty of tweakable parameters and a great sound make them hard to resist.

See the walkthroughs for coverage of key parameters with the Tape and Stereo Delay, plus an advanced example for creating dub-like delays.

Focus On - Sample Delay

With just few controls to play with, Sample Delay isn't a plug-in drenched in features, but don't let this apparent simplicity put you off. Beneath its relatively simple exterior lies a number of interesting practical possibilities covering both corrective and creative applications. One primary use is in the time alignment of microphone capsules, especially where microphones are placed at relatively large distances.

Try applying a delay of 129 samples to your close mics for every metre of distance between them and a set of ambient mics. You can also try delaying one side of a stereo signal by around 50 to 100 milliseconds, to recreate the Haas Effect. The result is a shifting of the stereo image, with the delayed version appearing slightly quieter to the ear.

Step-by-Step - Reverb Basics

1: Create a bus send from the synth line on the first track. Across this bus, load an instance of the Tape Delay plug-in. You can set the wet/dry ratio either by reducing the bus send level, or by lowering the aux fader's level.

2: As Tape Delay works with just a single tap, it's well worth thinking about the relative panning of the dry and wet signals. Panning each component slightly off-centre can help create some additional movement to the delay effect.

3: With the Tempo Sync button active, you can quickly move between a variety of tempo-matched delay settings. The dotted settings (like 1/8 dotted, which we're using here) often work well as the delays will fall on syncopated beats.

4: Adjust the Feedback control to create the desired amount of repeats. We're reducing it to 0%, as we just want a single delay, but you could also choose to raise Feedback right up until the delay goes into self-oscillation.

5: Change the colour of the repeats using the Low Cut and High Cut controls. Setting High Cut around 1600 Hz rolls off the high-end nicely, creating a darker delay. Use Low Cut to produce a progressively thinner sounding delay.

6: To change to degradation of the virtual tape unit, try varying the Clip Threshold and Flutter controls. Lowering the Clip Threshold will produce a more distorted sound, whereas Flutter adds small pitch deviances to the delayed signal.

This tutorial is endorsed by Point Blank. With courses in London, online and now in LA, Point Blank is the Global Music School. You can study sound to picture on their Music Production Diploma courses, with pro industry tutors.
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Logic Pro X Tutorial - Logic's Updated Delay Plug-ins

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With a graphic refresh and a host of new features, Logic's delay plug-ins have never looked or sounded so good. Mark Cousins experiences the joys of repetition... It's great to see a range of plug-ins receiving a much-needed facelift as part of the continual development of the application.

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With eight plug-ins covering distortion, Logic Pro X offers everything from soft saturation to aggressive fuzz. Mark Cousins gets crunchy...

Engineers have long used distortion as a means of adding 'mojo' to a sound - whether it's a touch of drive across some drums, or some aggressive distortion on a synth bass. Logic Pro X has a bewildering selection of plug-ins from the versatile Distortion and Overdrive plug-ins, to more guitar-friendly options like Pedalboard. Each plug-in offers a slightly different approach and characteristic of distortion.

Distortion can be highly effective applied as a subtle tool, adding drive and character to an input rather than it sounding notably distorted. In other examples, you'll want to use Distortion as a noticeable effect, where the sound is mangled to a point where it's (almost) unrecognisable from the original source.

Distinguishing between these two contrasting approaches, therefore, and the plug-ins you can use to achieve them, is a key component in using Logic Pro X's distortion plug-ins effectively.

For simple, relatively transparent applications of distortion, you'll want to use the Distortion, Overdrive, Distortion II and Bitcrusher plug-ins. These all benefit from a relatively simple control set, and when applied discreetly, don't necessarily result in the input being completely covered in distortion. Two initial stalwarts are Overdrive and Distortion.

On the whole, Overdrive is the subtler of the two plug-ins, producing a warm form of distortion associated with FETs. Distortion, on the other hand, models the distortion produced by solid-state devices, which tends to have a slightly harsher tone.

Harmonic sizzle

The process of distortion will, of course, add additional harmonic material to the input - adding extra body and an increased vibrancy in the high end. To rationalise this extra harmonic information, both plug-ins feature a post-distortion Tone control that works as a low-pass filter to tame the additional harmonic material.

On the whole, you'll probably want to contrast two different basic sounds - one that creates a warmer 'vintage' tone with a small amount of high-frequency tone attenuation and another that preserves the bright timbre by leaving the Tone in its open setting.

One of my favourite tools for adding drive and colour is the Bitcrusher plug-in, which models the characteristic distortion added by the analogue-to-digital (A/D) conversion process. Using a small amount of Drive combined with a lower Bit Rate (like 8-, 12- or 16-bit) instantly gives edge to a sound, but in a way that contrasts nicely with the analogue-like drive introduced by the Overdrive and Distortion plug-in. Pushed harder (with a noticeably grungy Bit Rate and some Downsampling introduced), Bitcrusher sounds anything but polite, turning a range of sounds into instant Chiptune.

We've now started to move beyond softly saturated tones and into the realms of distortion producing a deliberately mangled output. For noticeably distorted sounds, you might want to move beyond what Overdrive and Distortion alone can offer and look at other sources of distortion; like Amp Designer and Pedalboard, as well as using other plug-ins that complement and enhance the characteristics of distortion.

Plug-ins Distortion II is ported from the distortion section of the Vintage B3 Organ instrument, and is a good alternative to the analogue-like saturation of the Overdrive plug-in, particularly using the two Class AB Type settings. It's Tone control is based on a high-pass filter; Distortion's is based on a low pass.

Pedalboard's collection of Distortion pedals - like Rawk! And Monster Fuzz - delivers a rich, fuzzy distortion that sounds great on aggressive synths. Pedalboard is also a good place to explore the possibilities of tone stacking, where the distortion isn't applied as a single-stage effect, but instead uses a number of different pedals together.

Another useful feature of Pedalboard in respect to tone stacking is the ability to run two separate busses - A and B - so that the distortion can run in parallel. This is particularly useful for wide stereo distortion, where two different distortion channels are panned left and right respectively.

Designer distortion

Amp Designer creates distortion from a virtual amp head with additional speaker simulation. Much like the Tone control on the Distortion and Overdrive plug-ins, the speaker simulation can be used to tame the excessive high-frequency content created by distortion. Indeed, you can even use the Clean Preamp option as a means of using Amp Designer as the back end to a distortion chain either created with Pedalboard and/or individual distortion plug-ins.

Likewise, an Amp-only setting - without the speaker modelling - has a bright, crunchy distortion which is a good alternative to the Distortion and Overdrive plug-ins. For more extreme distortion effects, add a few effects before the distortion. Compression and EQ can have a big effect on the behavior of the distortion.

More plug-ins Phase Distortion has a more abstract sound, being based on a modulated delay line rather than a traditional non-linear amplifier. Phase Distortion's unique 'warbly' sound is a somewhat niche effect, although it can produce some interesting results on drums.

Compression, for example, changes its dynamic qualities, effectively providing more 'meat' for the distortion to work with. Likewise, EQ influences the tone of the distortion, and is particularly effective where the input is placed in a narrow part of the sound spectrum, producing a unique lo-fi treatment.

Logic Pro X's arsenal of distortion plug-ins, though overwhelming at first, offers a wealth of creative opportunity. Master them and you'll give your mix edge and drive without the original instruments being degraded. Likewise, extreme distortion - pushing plug-ins like Bitcrusher into more aggressive treatments, or tone-stacking multiple fuzz pedals in Pedalboard - can mangle instruments and produce levels of aggression and bite no other plug-in can deliver.

Step-by-step: Distortion options

1. Working with the drums on track 1, insert an instance of the Distortion plug-in. For a simple distortion effect, position the Output to -6dB, set Level Compensation to ON and adjust Drive to get the desired amount of extra crunch.

2. Distortion creates lots of added harmonic material, dependent on the amount of Drive. By default, Tone attenuates a proportion of the harmonics, but by increasing it to 20,000Hz, you hear the distortion in its full 'brightness'.

3. Try placing an instance of Overdrive on the same channel, matching the settings between the two plug-ins. A/B the two flavours of distortion - note how Overdrive has warmer sound, while Distortion sounds brighter and edgier.

4. Try listening to the sound of Clip Distortion. To understand how it works, start by creating an initialised 'reset' - with Mix at 100%, Drive at 0dB, the two filters (Tone and Clip Filter) at extremes and the LP Filter at 20,000Hz.

5. Clip Distortion offers the unique opportunity to filter the distorted element of the signal. Here, we filter the input to focus on the snare (70 Hz and 6600Hz for Tone and Clip Filter respectively) and increase the Drive to around 16dB.

6. Use Mix to create a blend between the clean signal and the frequency-focused distortion. Set the Output Gain to around +7.0dB and A/B, noting the Clip Distortion's colour of the midrange without sounding notably distorted.

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Learn How To Create Amazing Distortion Sounds In Logic Pro X

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With eight plug-ins covering distortion, Logic Pro X offers everything from soft saturation to aggressive fuzz. Mark Cousins gets crunchy... Engineers have long used distortion as a means of adding 'mojo' to a sound - whether it's a touch of drive across some drums, or some aggressive distortion on a synth bass.

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It's been over a year since we last did freeware, so we're kicking off our freeware 2018 roundup with our favourite 7 synths available to download today. (Click the titles to download)
Platform: Mac, PC
Platform: Mac, PC
Platform: Mac, PC
Platform: Mac, PC
Platform: Mac, PC

TAL makes some great freeware and Noisemaker (actually NoiseM4k3r) is an update to the rather great Elek7ro synth that we have often included in our synth roundups. It's a three oscillator wavetable synth with up to six voices and a stack of great effects - reverb, delay and bitcrusher - that lift its sounds well above most. And you get a whopping 256 of those to explore, too.
Platform: Mac, PC

Sinnah has just been upgraded to v1.1. The instrument includes 50 free presets and is a complex wavetable synth capable of producing many and varied sounds compared to your bog standard virtual analogue (of which it is one). You might have to work with it to get the best from it but the potential is certainly there including a fantastic Delay Matrix.

We love U-He software and this is one of the company's classic synths. It came out a decade ago but runs like a dream on modern DAWs as U-He has kept it well up to date, the last big revision being in 2015. It uses various forms of comb filter as its synthesis source which means it sounds a bit different from most of what you'll find out there in freeware land with lots of percussive and delay-type presets.

We've always included Crystal in our Best Of freeware lists and probably always will. Originally released in 2002, its developer has kept it up to date, so it runs in more modern DAW setups. It looks a little dated now, but still sounds pretty fresh and vibrant, even next to a lot of our paid-for plug-ins. There are rumours that post-nuclear war, only rats will survive. If that's true, they'll be using Crystal.

A lot of these synths have been around for a while to almost attain classic freeware status but Helm is a newer kid on the block, certainly in terms of its looks anyway. Its GUI is quite superb with lots of click and drag graphics to really change the on-board sounds although you might not want to as what you get is a bunch of presets that are certainly among the most interesting you'll find.

Synth1 is another freeware synth classic that we have to include every year, or people will march on the MT offices with virtual pitchforks. It's apparently modelled on a Nord Lead 2 (sonically, not visually).

We managed to get it working in the latest Logic pretty much straight away, although issues are creeping in, like odd mouse movements. Worth persevering with for the sound, but possibly its last inclusion (sound of protest march starts - oh dear).

The Yamaha DX7 was one of the most popular synths of the 80s. Trouble is,
it was a pig to program, and the restrictions of hardware made it even harder. Dexed tries to make the process easier within software and succeeds, bringing real-time control to many parameters. There are 32 typical FM presets - chimes, fast-attack sounds and keys - making this a distinctive synth in a sea of virtual- analogue wannabes.

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Fantastic Freeware 2018: The 7 Best Freeware Synths

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It's been over a year since we last did freeware, so we're kicking off our freeware 2018 roundup with our favourite 7 synths available to download today.

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10 tips for creating better mixes from iZotope

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10 tips for creating better mixes from iZotope

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Welcome to iZotope's free online mixing masterclass! These 10 tips from award-winning mixer Enrique Gonzalez Müller and iZotope's Education Director Jonathan Wyner are split across the five video lectures below.

Week 1: Developing good mix habits

All the tips in this course are geared towards developing good mix habits that you can bring to every project you work on. These first two tips can help you prepare yourself and the session file for the creative work of mixing.

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Welcome to iZotope's free online mixing masterclass! These 10 tips from award-winning mixer Enrique Gonzalez Müller and iZotope's Education Director Jonathan Wyner are split across the five video lectures below. All the tips in this course are geared towards developing good mix habits that you can bring to every project you work on.

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Use Flex editing to adjust the timing of groups of drums tracks while maintaining phase accuracy among the tracks.
Create a Group
First create a group for the drum tracks. You should do this before you record the tracks, but you can do this afterward if necessary. However, you need to create the group before you start editing the drum tracks. To create the group with these steps.

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1. Click inside the group slot in one of the drum track channel strips.
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2. Choose a new group from the drop-down menu.
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3. The Group Settings panel will open in the Inspector. Click to enable the options for Editing, Phase-Locked Audio, and Record. It's also a good idea to give the the group a descriptive name.
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4. Close the Group Settings, then Option-click the group slot in each of the other drum track channel strips on the Mixer to assign them to the group.
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Record and Edit
Record the drum tracks. When you're done recording, use Quick Swipe comping to create the best composite take. Make any other edits, such as copying sections from one part of the song to another, and so on.
Flatten and merge
When you're satisfied with your comping and editing, flatten and merge the take folders with these steps.

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1. Open the Take Folder pop-up menu in the upper left corner of one of the take folders.
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2. Choose "Flatten and Merge" from the menu.
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Note: This is recommended because phase-locked editing requires that all tracks in the group have the same start position and be exactly the same length. By merging all the regions and takes on each track to one audio file, discrepancies are less likely to appear. Also, because Logic's transient detection is file based, you need to go through the process of adjusting transients only once for each Q-Reference track.
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Choose tracks as Q-Reference
Decide which tracks should be used as Q-Reference tracks that will be used to determine quantization for the group. Typically, a snare and kick drum track are used as reference tracks. If you used multiple microphones on either drum, choose one track for each.

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1. Open one of the tracks you want to use as a reference in the Audio File Editor window and choose Audio File > Detect Transients.

2. After Logic analyzes the file, check the transients in the Audio File Editor to make sure they are positioned correctly. Use the plus and minus buttons to add or subtract transients, move incorrectly placed transients, or manually add or remove transients with the pencil or eraser tools.

3. Repeat these steps for any other files you will use as Q-Reference.
&NewLine;&Tab;Note: It's not necessary to adjust the transients for the other tracks in the group&semi; their timing will be adjusted with sample accuracy based on the transients in the Q-Referenced tracks.

4. In the Track Header of the Tracks area, enable the Q-Reference button for the tracks to be used as timing reference.
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Note: If you need to return to the Audio File Editor window to adjust the transients for any of the tracks after this stage, it's very important to disable the Q-Reference buttons for all tracks in the group first, otherwise, the transient edits will not carry over to your subsequent timing adjustments.
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5. In the Track Inspector for one of the tracks, click the Flex drop-down menu and choose one of the Flex Modes. For drums, Slicing is usually the best choice, but feel free to experiment with the others. The Flex Mode will be assigned to all tracks in the group.
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Enable Flex View in the Tracks area

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1. Click the local Edit menu in Tracks area.

2. Choose Show Flex Pitch/Time from the drop-down menu.
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Apply Timing Adjustments
Perform your timing adjustments. For example, apply quantization to the tracks, or manually insert flex markers and drag audio in the reference tracks to desired positions.

When you apply quantization, you may find that some transients are quantized to unexpected positions. For example, if you choose a quantize value of 1/8 note, you may find that in places where there is no transient on the 1/8 note grid, a transient in between grid positions, such as an in-between 1/16 note is moved to to the nearest 1/8 note. You can prevent this by setting an appropriate Q-Range value in the region parameter box:

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1. Click the Advanced Quantization disclosure triangle at the bottom of the region parameter box.

2. Click the up/down arrows to the right of the Q-Range parameter.

3. Choose a relatively short value, such as 1/24 or 1/32. Experiment until you get the result you want.
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Note: Q-Range limits quantization to transients that fall within the value you choose. For example, if you choose 1/32, then any transient farther away than 1/32 note from the quantization grid you choose will not be quantized.

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Logic Pro X: Tips for phase-locked drum edits

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Use Flex editing to adjust the timing of groups of drums tracks while maintaining phase accuracy among the tracks. Create a Group First create a group for the drum tracks. You should do this before you record the tracks, but you can do this afterward if necessary.

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We're all different as engineers and music producers. Amongst our vast array of differences, however, there are some striking similarities. We all cringe at the word "exposure", we all eventually become coffee addicts and we all go through a steep learning curve when it comes to reverb.

I am going to go as far as to admit that there may have been a time in my life when I thought a "Hall" reverb referred to a hallway. What? It's a very understandable mistake if you have no idea what a hallway sounds like.

If you're somewhat new to the production world (and maybe even if you aren't), popping open that reverb algorithm can be a bit daunting. Even if we just ignore the finer controls, just knowing what different "types" of reverbs mean and what to expect from them can be confusing.

Fortunately, you've Googled your way here and I can help!

The basic idea is this: a reverb processor is a device that uses an algorithm (magical math) to turn a source sound into the reverberant version of that sound. That algorithm can be configured in a large number of ways, but the designers generally like to set up some basic configurations that imitate the typical reverbs that we frequently come across. Customarily, the reverb types are Rooms, Halls, Chambers, Plates, and Ambiences.

The fun part is that every algorithm is different and every designer's interpretation of how to imitate each type is completely subjective. The good news is that there are some commonalities that are generally agreed upon that can help hedge our expectations.

1. Room Reverb

Our most basic reverb type is a room. A room is a space with walls. Doesn't get much simpler than that. You are probably reading this article while you are in a room right now. Yup, take a good look around, that's a room.

Rooms come in many shapes and sizes. Your living room is one room, the central space of a cathedral is another room and an underground parking garage - also a room. Your living room is probably fairly small and walled in wood and/or drywall. The cathedral is probably pretty friggin' big and might be made out of marble. The parking garage is probably wide, with a low ceiling and made of concrete. What you'll notice is that all of these spaces sound completely different. But, there are a few similarities.

The first is that rooms are designed for shelter and for people to exist, but not specifically for great sound. As a result, unless you happen to be in a Frank Loyd Wright house, rooms tend to have practical shapes such as flat surfaces and parallel walls. These flat, parallel, reflective boundaries cause sonic anomalies such as modal interference, standing waves, flutter echo, rings and resonances. Simply put - they tend to have a distinct colored character.

There's a lot of "imperfection" in the sound of a room - but it's exactly those imperfections that can make them compelling. Let's be honest, who hasn't wanted to record drums in a parking garage? And your friend's acoustic guitar might actually sound pretty good in that living room.

Rooms give a natural color and liveliness to a sound. They also sound like a more intimate space and can give the impression that all of the music is being played right in front of you. When we're going for a natural, in-person, live kind of sound (a pretty common goal in acoustic styles of music like Folk and Jazz), room reverb is a good choice.

2. Hall Reverb

I mentioned that rooms are designed for practical purposes and not with the goal of fantastic sound in mind. What if I told you there are rooms out there that are designed exactly with the goal of incredible sound in mind?

Concert Halls are spaces that are dedicated to the enjoyment of music. They're built to specifically have minimal room modes, echoes, rings, etc., and intended to sound as tonally even as possible (although some are built to amplify lower frequencies to give an orchestra a more grandiose sound). And many halls are built to help propagate the sound, which is why many halls are bow shaped.

Halls are the sound of class. Diamond sequins, Stradivarius violins, a Fazioli pianoforte and possibly Barbara Streisand would all be at home in a concert hall. That Folk band I was talking about in the previous segment, not so much. They're the salt-of-the-earth, homespun people who don't see much use in putting on airs. So, they're happy with vibey room reverb. Celine Dion, she likes a perfect, lush sounding hall.

3. Chamber Reverb

Chambers are kind of the opposite end of the spectrum from halls.

Once upon a time, before digital reverb units, there lived a humble recording studio owner. And this studio owner wanted the sound of a vast reverberant space. Unfortunately, real estate being expensive, this studio owner did not have the resources for a giant hall. So he dedicated a smaller room with very reflective surfaces and preferably oblong angles to the task. Because this smaller room was especially reflective, the reverb tailed on and on when sound was ported in through a speaker. Unfortunately, because the room was small and full of dense echoes, the room had a particularly uneven timbre. Fortunately, people like quirky, vibey stuff.

And thus, the Chamber reverb was born.

Now admittedly, that story is a bit embellished. There's obviously no such thing as a humble studio owner - but that's more or less how it happened. Chambers have a ton of color and texture and can sound great on all sorts of sources. Anything that needs a distinct bump in energy or extra texture can benefit. Chambers were largely used on older R&B and Classic Rock records, so if that's the vibe you're going for - fire up that reverb.

4. Plate Reverb

And this brings us to one of the most breathtaking reverb sounds of them all: a sheet of metal hanging in a box.

Ok, that's a bit of an understatement - it's more than that. There's also a tiny driver and two pickups. I really feel like I should be saying more, but there isn't much more to say.

There are some interesting qualities to plate reverbs. A sheet of metal is a two-dimensional space. In a three-dimensional space, there are discrete echoes at the front of the reverb tail and as the reverb tails out the echo density increases. In a two-dimensional sheet, the echoes are the same density from the start of reverberation to the end.

Because the speed of sound is much faster in metal than air, plates have a higher "echo" density. These two factors are what give plates their signature "smooth" reverb tails. Lastly, the higher frequency tones exhaust their energy much faster than lower tones, while lower tones actually take longer to build up. This creates the effect of higher frequencies living at the front of the reverb tail while lower frequencies tail out later on. In other words: metal doesn't just look shiny, it also sounds shiny.

This all makes plates useful for a wide number of sounds. Because of the even distribution of echoes in the tail, plates tend to get along very nicely with other reverbs.

If you happen to have a drum kit, with a couple of room mics, you can probably sneak a plate reverb onto the snare and toms and still have a cohesive sounding kit. This may not be as true if you were to throw an additional room reverb on the snare. Then you get these two different rooms at the same time sounding like the plot of Inception to your ear. Plates are also great at bringing a subtle "pop" or brightness to a sound source, making them ideal for pop vocals, acoustic guitars and really anything else for that matter.

5. Ambiences

Ambiences are the real deal when it comes to digital algorithms. The reality is that an algorithm isn't a room, hall or a plate. An algorithm is its own thing. Just like how a chamber isn't really a hall and a plate isn't a room.

Algorithms sound like what they emulate, but not the same. The word ambience really just refers to ambiguous space around something. In my mind, this is actually where reverb algorithms shine.

Conventionally, ambiences will refer to very short reverbs with any sort of setup the designer deems worthy. That can mean a reverb that's all early-reflections, late-reflections, highly diffuse, highly non-diffuse, whatever - as long as it's indistinct.

While that might sound unexciting, it's actually perfect for sounds that already have a ton of character and dense harmonics like synths. The shortest ambiences are great for any source that you want to sound dry, but still have a sense of three-dimensionality (like rap vocals).

Bonus: Non-Linear Reverb

There's also a style of reverb exclusive to the digital domain called a Non-Linear Reverb. Like Ambiences, these reverbs are sort of non-categorizable but with one key difference. In all other forms of reverb, as the tail rings out energy is dissipated and the echoes become quieter and quieter.

In a Non-Linear reverb, the echoes can swell up before tailing out or even crescendo until the very last set of echoes. This doesn't happen in reality, but can be a very cool effect in the musical world.

Non-Linear reverbs can create a rhythmic feel when returned from drums, act like a slapback on guitars and work well on sounds that you might want to keep in-your-face while still having a distinct reverb sound on them.

Mixing with Reverb Course

To learn more about how you can create an exciting, evocative, three-dimensional mix using reverb, check out our new Mixing with Reverb course, 6+ hours of in-depth training that will help you take your mixing skills to the next level.

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We're all different as engineers and music producers. Amongst our vast array of differences, however, there are some striking similarities. We all cringe at the word "exposure", we all eventually become coffee addicts and we all go through a steep learning curve when it comes to reverb.

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Reverb is vital. Songs with reverb tend to stand out when it's used effectively. Reverb is such a useful audio effect because it's incredibly flexible.

Why do those vocals sound so heavenly? How is that guitar tone so dreamy? Why do I feel like I'm being transported to an actual space where these instruments exist??

This audio effect is as much a technical tool, as it as a creative one. Reverb creates emotion, a vintage feel, space, depth and even punch (check out no. 7)! It's used to craft both natural sounding spaces and totally surreal atmospheres.

So to celebrate the tracks that refuse to take reverb lightly, here are 8 tracks of different genres that use reverb in masterful and original ways. Get inspired, then hit the knobs!

1. King Tubby "Dub You Can Feel"

Osbourne Ruddock a.k.a. King Tubby was a key figure in the Jamaican soundsystem culture of the 50s.

His production and sound engineering work was pivotal to the development of dub and reggae in the 60s and 70s. He's well known for his innovative technical work and for essentially inventing the concept of the remix.

Reverb is one of Tubby's signature secret weapons-and what makes dub sound so special. Jamaican 45s often press the main track on one side, and an instrumental "version" on the other.

Reverb is one of Tubby's signature secret weapons-and what makes dub sound so special.

When producing a 'version' without the vocal track, Tubby would take a lot of creative liberty. He'd accentuate this or that instrument, add some delays, echoes and reverb-a major innovation at the time. This would soon be known as remixing.

A perfect example of Tubby's deft touch for reverb is present on the track "Dub You Can Feel." Listen for the reverb on the snare - it's the classic spring reverb "boing" effect common in dub music.

2. Joe Meek "I Hear A New World"

Joe Meek is a notable British sound engineer, record producer and songwriter active in the 50s and 60s. His experiments with effects like overdubbing and sampling are also foundational to space age pop and psychedelic outsider music.

In his cult classic album I Hear a New World from 1960, Meek uses various types of effects including pitch shifting, panning and natural reverbs.

One of Meek's most famous techniques was using different reverberant spaces in his studio and house for recording instruments with natural room sound-like under his staircase or in his bathroom.

Listen to the different kinds of spaces he creates with various reverbs, especially the uniquely spacey reverb on the backup vocal line: 'I Hear a New World.'

3. Led Zeppelin "When The Levee Breaks"

This Led Zeppelin song masterfully uses multiple effects.

The drum performance is central to the track. It was famously recorded on a brand new Ludwig kit in the hallway of a three story staircase at Headley Grange. Two Beyerdynamic M160 microphones were placed at the top of the stairwell for the recording. This gave it its distinctive natural sounding 'big room' reverb and muffled sound that makes it standout from other drum recordings.

Listen to the end of the song for the famous 'swirly' effect achieved with a reversed harmonica, a backwards echo, phasing, flanging and panning. This gives the impression that all the instruments are spinning around Robert Plant's vocals which remain in the middle.

4. Grimes "Venus In Fleurs"

Reverb and vocals work well together, especially in the ethereal dream-pop side of things.There's no better way to add a dramatic, angelic effect to a vocal track.

Canadian producer, songwriter and vocalist Grimes has undoubtedly popularized this style of vocals in experimental pop of the 2010s. For this approach to production, understanding what the lyrics say isn't the focus. It's all about the feeling and the ambiance created by the vocal melody and reverb.

It's all about the feeling and the ambiance created by the vocal melody and reverb.

In 'Venus In Fleurs,' the reverb places the vocals in the background, almost as a far away ambient layer. The reverb on the drone guitar adds to the washed out distant effect.

Grimes epitomizes a generation of bedroom producers from the last decade who've taught themselves the ins and outs of DAWs, intentionally pushing effects to the max and creating their own stylistic mark. This is the punk approach to pop!

5. The Drums "Down By The Water"

The sound of Indie Pop band The Drums is modern mix of The Beach Boys, The Smiths and Joy Division. Their track "Down by the Water" is an especially beautiful example of what reverb can do for a performance.

Everything in "Down By the Water" is essentially drenched in an enveloping hall reverb that glues it all together. The transients of the snare, tambourine and vocals linger with a dreamy, intentionally muddy quality. This gives the track a vintage sound and intensifies the emotion of the song. Listen to how the song ends: a long reverb tail lingers until it fades and comes back reversed.

Everything in "Down By the Water" is essentially in an enveloping hall reverb that glues it all together.

This song exemplifies a certain kind of indie pop sound from the 2010s which revisits New Wave and gives it a dream pop quality, soaking up your tears with 100% wetness.

6. Shinedoe "Cosmic"

Chinedum Nwosu a.k.a. Shinedoe is a DJ and producer based in Amsterdam. Channeling the melodic richness of Detroit techno and Chicago house, she crafts lush tracks that create a sense of depth and weight.

With clear dub influences in its rich reverb, the track "Cosmic" makes an interesting use of the effect on synths pads, vocal samples and claps. The decay extends the harmonic content of chords, turning them into an enveloping ambient layer. It glues multiple synths together and creates a cohesive whole.

The decay extends the harmonic content of chords, turning them into an enveloping ambient layer.

In the club, reverb is the purveyor of transformative and enveloping club experiences-from dub techno to psytrance. Nice reverb heard on a Funktion One? Anytime.

7. Phil Collins "In The Air Tonight"

This chart-topping pop hit by British singer-songwriter and drummer Phil Collins is the most famous example of gated reverb. This type of reverb has a very short decay trail that enlists the help of a noise gate.

The gated reverb heard on "In the Air tonight" gives a punchy, big sound (listen for it on the snare and drums) that remains tight and clean.

The cool thing about gated reverb is that it doesn't try to replicate a natural sounding effect-It would be impossible to find a reverb like this in a real space.

Instead, it lends epic drum solos incredible power. Thanks Phil, for giving us the most melodramatic drum break ever.

8. Blake Mills "Shed Your Head"

Blake Mills is a guitarist, songwriter and GRAMMY award winning producer. He has credits on albums by the like of Sky Ferreira, Fiona Apple, Alabama Shakes, John Legend and Perfume Genius.

The track "Shed Your Head" from Mill's solo record Heigh Ho is an impressive example of re-creating realistic space with organic reverb. The reverb on the song creates convincing room subtlety.

The reverb in this track places all the instruments in a natural sounding space, adding a warm texture to the entire performance.

It goes to show that reverb does not always need to be blown out or excessively wet for effect.

Mill's guitar tone sounds so much richer and velvetier with that reverb on it. The reverb in this track places all the instruments in a natural sounding space, adding a warm texture to the entire performance.

As Mills recounts in an interview "Guitar has such a history of flashy playing. I was impressed by it too..."

But after spending more time with records by Nina Simone and Thelonious Monk, he soon dropped his desire to be an 'impressive' player: "I just wanted to be a moving player."

Could part of this magic also be the right reverb?

Many Shades of Reverb

As these examples demonstrate, reverb is extremely versatile-whether you're making techno, rock, pop or experimental music.

Reverb knows no genre boundaries and it's as much a technical tool as it is a creative one. It's the ultimate shaper of moods and spaces.

Get inspired by these songs, artists and their unique take on reverb to test out all the flavours of the effect in your own productions.

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8 Songs that Belong in the Reverb Hall of Fame | LANDR Blog

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Reverb is vital. Songs with reverb tend to stand out when it's used effectively. Reverb is such a useful audio effect because it's incredibly flexible. Why do those vocals sound so heavenly? How is that guitar tone so dreamy? Why do I feel like I'm being transported to an actual space where these instruments exist??

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Twice a month we revisit some of our reader favorite posts from throughout the history of Audiotuts+. This tutorial was first published in July 2008.

When it comes to special effects, reverse reverbs are amongst the most useable processes there are. Although the method for creating these effects is often looked upon as complex, the right technique can lead to a transparent workflow and impressive end results. In this tutorial we'll apply a reverse reverb effect to a vocal--a technique you might have heard used at the beginning of Justice's 'DVNO'.

Note: this article contains embedded audio that will not display in a feed reader. Click back to the site to read the tutorial with audio, or download the audio files at the bottom of the post.

Step 1

Choose the file you want to process and import it onto a fresh track in your DAW. It's important to ensure this initial sound is reasonably dry and unprocessed. This way, any further processing will be easily distinguished and well defined. Make sure to trim any clicks or other artifacts from the start and end of the file, as again, this will ensure a cleaner end result.
dry_vocal.mp3
As you can see from the screenshots, I have used Cubase 4 here, but this technique can quite easily be repeated in other applications by using the same techniques.

Step 2

Insert a good quality reverb processor onto a return or buss channel in the same way as you would in a standard send/return configuration. I have used Steinberg's 'Roomworks' plug-in here but any studio-quality reverb unit can be used.

Load a preset with a very long decay time and little to no damping. Of course, you can program the reverb yourself if you feel confident. As for as the parameters needed, you can afford to use something a little over the top here in every area. This is a special effect so don't feel constricted by the usual rules of thumb you might apply.

Step 3

Send a large amount of signal from your chosen sound to this new reverb buss. To do this, simply use an auxiliary send and dial in a high gain amount. Again, feel free to use higher values than you would usually.

Be sure that you avoid clipping at any stage here so that you don't introduce any unwanted digital distortion into the chain.

Step 4

Reverse the sound you originally imported. This can be achieved directly in the arrange window of some DAWs (e.g. Steinberg's Cubase, Apple Logic Pro, Ableton Live) or you may wish to use a standalone audio editing application.

It's a sensible idea at this point to make a copy of your file so that you have quick access to both the edited and original versions. On playback you should now hear your reversed file being effected by the heavy reverb setting.

Step 5

The next step is to export or record the sound of the reverb alone. The method you use here will depend on the DAW you are using, but generally the routes available to you at this stage are very similar.

In most DAWs you are able to route a channel to 'No output' or 'No Buss'. This essentially means the files are still playing back but the resulting output is not heard through you audio interface outputs. The aux sends from this channel will still work and therefore you should only hear the reverb from the effects return on playback.

You are now able to export the area as usual and the file will only contain the reverb signal.

Step 6

Now re-import this 'reverb only' file you have created into its own new audio channel. Playback the file to check that it contains only the audio you need and then name the new file and the track to keep things organised.

Step 7

Again, reverse this file the way you did in step 4, using either the editing capabilities of your chosen DAW or an external audio editing application.

At this point you can also reverse the original audio file so it plays back as it did originally. This can be achieved using the method above or using the original or backed up audio file. On playing back the two channels individually you should now hear the original file and the reversed reverb.
reverse_reverb.mp3
Step 8

At this point you need to start thinking about making the two files work together so that they playback seamlessly in an organic way. The first obstacle here is their timing, so switch off any auto snap mode your sequencer may have and start to move the two files around.

You should be able to quickly find points in the reverb file that resemble points in the original sound. Using these as anchors you can line up the two files so that the rise of the reverse reverb leads into your original audio perfectly.
final_processed_vocal.mp3
Step 9

Once the placement of your audio is sorted you can start to fine tune things. Start by applying fades to the start and end points of all your files. This will ensure smooth playback of the whole sequence with no clicks and pops.

Step 10

Finally, use your DAW's mixer to balance the level of the two files so that they compliment each other in your mix. Also, try adding a small amount of the reverb to the original audio file to give it some space and help it blend with the reverse effect. If you like, at this point you can bounce the whole sequence into one file to make things easier.

For some interesting results, why don't you try processing the reverse reverb audio further with some auto pan or tremolo effects? You can even apply further delay to the reverse effect for a really spacey feel.

Listen to an example of this technique in use:
example_of_use.mp3
Download the Play Package for this post (11.2MB)

Contents:

Printable PDF Tutorial

Audio files

Mo's Cubase project

Hear examples of Mo's work or drop him a line at www.myspace.com/mohawkmuzik or www.MoVolans.com.

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Twice a month we revisit some of our reader favorite posts from throughout the history of Audiotuts+. This tutorial was first published in July 2008. When it comes to special effects, reverse reverbs are amongst the most useable processes there are.

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It's time for our annual MusicTech freeware roundup! We've gathered the best free software titles in a variety of music-production categories and presented them here. Want to make music for nothing? Look no further...
Last year, we predicted that this would become an annual event here at MusicTech, and we were right! So successful was 2015's Freeware 50 that we've decided to do it again for 2016 - looking forward to 2017, that is, list the best 50 freeware plug-ins in the world, right now!

The freeware scene has been around since the birth of software. It highlights either free versions of paid-for apps (these might act as an advert for a more fully featured version), or complete fully functioning software from a small company which programs for fun. There are a huge number of developers - from big names like Native Instruments right down to one-person companies.

We admit that we have some favourites, but this year we've kept a tight rein on things, with firm rules for our list of the best free software out there. These are:

1. It's an-all new list, apart from half a dozen that just had to be included again and, yes, Crystal is one of them.

2. We've included both real and synth instruments; effects (creative, mixing and mastering); plus other studio production options, including utilities.

3. We've also chosen all (bar one!) titles that are available on Mac or PC. So, sit back and enjoy MusicTech's freeware of 2016... and don't spend a penny on any of it.

So without further ado, let's begin our Freeware roundup. Click on any of the link buttons below for each individual category (With download links)

2015 Freeware Roundup

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The Best Freeware for Music Makers 2017 - The Complete Collection

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It's time for our annual MusicTech freeware roundup! We've gathered the best free software titles in a variety of music-production categories and presented them here. Want to make music for nothing? Look no further... Last year, we predicted that this would become an annual event here at MusicTech, and we were right!

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Finding solid emulations of real/acoustic instruments is a tricky endeavour, one made worse when you want to spend nothing. Fortunately, we've 7 great examples. (Click the titles to download).
Type Mac, PC, VST

We've included Ticky Clav in pretty much every freeware roundup we've ever done but decided to put the whole Big Tick collection forward this time round because, as you have to register to download one, you may as well get the lot while you're there. It includes EP-Piano, Angelina formant and Rainbow Hybrid synths; and the rather great string machine emulator Cheezemachine.

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Fantastic Freeware 2018: The 7 Best Freeware Real/Acoustic Instruments

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Finding solid emulations of real/acoustic instruments is a tricky endeavour, one made worse when you want to spend nothing. Fortunately, we've 7 great examples. (Click the titles to download).

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Are your productions in need of some decent after-effects that don't come with a price tag? Try out these 7 freeware effects. (Click each title to download).
Type Mac, PC, VST

We've got a good workhorse EQ so how about a decent vintage one to go with it? And actually PTEQ-X is not just a Pultec EQ clone - which is cool enough in itself - but has three types of EQ and four valve types, so offers your productions a lot of sonic variety. And as with so many freeware effects, the GUI is damn fine too. The Ignite Amps website is down as we write this but head on over to KVR for the download.

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Fantastic Freeware 2018: The 7 Best Freeware Effects

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Are your productions in need of some decent after-effects that don't come with a price tag? Try out these 7 freeware effects. (Click each title to download). Type Mac, PC, VST We've got a good workhorse EQ so how about a decent vintage one to go with it?

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Though we've only included one new addition in this final roundup, these 4 freeware utilities are too good not to share...
Type Mac, PC, VST/RTAS

PSP has a great reputation in the world of plug-ins and this analogue-style VU meter is the only freebie they do, so is well worth grabbing, especially if you don't trust your DAW's meters.

Type Mac, PC, VST/RTAS/AU

Blue Cat Audio does loads of great plug-ins and this freeware analyser allows real-time monitoring of your sonic frequencies. You can even customise the look of it, if you fancy.

Type Mac, PC, VST/AU/AAX

Proper mono-to-stereo requires a lot of thought, but this little freebie does it all for you, using a mid/side coding technique. It even pushes into surround territory.

Type Mac, PC, VST

You might not have realised it, but you need a bigger master fader - so it's a good job those folk at Sonalksis have realised this is the case, and come up with this for super-accurate monitoring within your chosen DAW.

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Fantastic Freeware 2018: The 4 Best Freeware Utilities

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Though we've only included one new addition in this final roundup, these 4 freeware utilities are too good not to share... Type Mac, PC, VST/RTAS PSP has a great reputation in the world of plug-ins and this analogue-style VU meter is the only freebie they do, so is well worth grabbing, especially if you don't trust your DAW's meters.

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Type PC, VST

It's amazing that Roland's TR-808 drum machine is still so popular since it made a comeback to kickstart acid house 40 years ago. There are plenty of software 808s around but this is as flexible as most and offers 15 classic drum parts to twist and tweak with controls like tuning, tone and decay knobs as well as velocity, level, pan and selectable output. It sounds better than it looks too.

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Fantastic Freeware 2018: The 7 Best Freeware Beat Makers

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Type PC, VST It's amazing that Roland's TR-808 drum machine is still so popular since it made a comeback to kickstart acid house 40 years ago. There are plenty of software 808s around but this is as flexible as most and offers 15 classic drum parts to twist and tweak with controls like tuning, tone and decay knobs as well as velocity, level, pan and selectable output.

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It's been over a year since we last did freeware, so we're kicking off our freeware 2018 roundup with our favourite 7 synths available to download today. (Click the titles to download)
Platform: Mac, PC
Platform: Mac, PC
Platform: Mac, PC
Platform: Mac, PC
Platform: Mac, PC

TAL makes some great freeware and Noisemaker (actually NoiseM4k3r) is an update to the rather great Elek7ro synth that we have often included in our synth roundups. It's a three oscillator wavetable synth with up to six voices and a stack of great effects - reverb, delay and bitcrusher - that lift its sounds well above most. And you get a whopping 256 of those to explore, too.
Platform: Mac, PC

Sinnah has just been upgraded to v1.1. The instrument includes 50 free presets and is a complex wavetable synth capable of producing many and varied sounds compared to your bog standard virtual analogue (of which it is one). You might have to work with it to get the best from it but the potential is certainly there including a fantastic Delay Matrix.

We love U-He software and this is one of the company's classic synths. It came out a decade ago but runs like a dream on modern DAWs as U-He has kept it well up to date, the last big revision being in 2015. It uses various forms of comb filter as its synthesis source which means it sounds a bit different from most of what you'll find out there in freeware land with lots of percussive and delay-type presets.

We've always included Crystal in our Best Of freeware lists and probably always will. Originally released in 2002, its developer has kept it up to date, so it runs in more modern DAW setups. It looks a little dated now, but still sounds pretty fresh and vibrant, even next to a lot of our paid-for plug-ins. There are rumours that post-nuclear war, only rats will survive. If that's true, they'll be using Crystal.

A lot of these synths have been around for a while to almost attain classic freeware status but Helm is a newer kid on the block, certainly in terms of its looks anyway. Its GUI is quite superb with lots of click and drag graphics to really change the on-board sounds although you might not want to as what you get is a bunch of presets that are certainly among the most interesting you'll find.

Synth1 is another freeware synth classic that we have to include every year, or people will march on the MT offices with virtual pitchforks. It's apparently modelled on a Nord Lead 2 (sonically, not visually).

We managed to get it working in the latest Logic pretty much straight away, although issues are creeping in, like odd mouse movements. Worth persevering with for the sound, but possibly its last inclusion (sound of protest march starts - oh dear).

The Yamaha DX7 was one of the most popular synths of the 80s. Trouble is,
it was a pig to program, and the restrictions of hardware made it even harder. Dexed tries to make the process easier within software and succeeds, bringing real-time control to many parameters. There are 32 typical FM presets - chimes, fast-attack sounds and keys - making this a distinctive synth in a sea of virtual- analogue wannabes.

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Fantastic Freeware 2018: The 7 Best Freeware Synths

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It's been over a year since we last did freeware, so we're kicking off our freeware 2018 roundup with our favourite 7 synths available to download today.

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Free drum sample packs - Producers Choice

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Free drum sample packs - Producers Choice

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Sample packs are essential for any frugal producer. Especially when they're free.

They capture all the analogue goodness without the price tag. No more choosing between buying food or saving up for that TR-808.

Free sample packs make you a better producer-because a musician is only as good as their tools right?

Here's my favourite insider spots to get the sample packs you need.

The Legowelt Sample Kits

Danny Wolfers (aka Legowelt) is a production genius. After watching him produce it's hard to believe that he thinks about anything else.

Fortunately for us, Wolfers offers free samples from some of the most iconic (and expensive) instruments. Incredible.

Our Favourite: The Yamaha DX-Files

Download them HERE for free.

Wavy.audio Free Sample Packs

Wavy.audio offers a new and diverse sample pack every week. Completely free.

Their subscription service allows you to subscribe to their mailing list and get every new sample pack to your inbox as soon as it's released.

You can even submit your own pack and get it featured on the front page with a link to your site.

Grab all the packs HERE for free

Ground Your Sound

If you wanna ground your sound in some seriously superb samples then the free Ground Your Sound sample pack from Cristofer Odqvist and The Pro Audio Files is a must-have.

There's all sorts of useful tools in it-from "Destroyed Drum Machines" to a collection of reverb tails that are perfect for layering.

Grab the pack HERE and file it under Pro.

Preview the free top-notch sounds below:

Liveschool Drum Samples & Racks for Ableton

Liveschool offers some great resources for artists including a huge amount of free samples.

You have to give a bit of info. But once you're in, there's tons to choose from.

Grab them HERE.

Our Favourite: Vermona DRM1 Drum Rack

Ableton Free Packs

Of course it makes sense for a DAW to provide some amazing sample packs. Ableton does just that with their Free Packs section.

You can find a whole bunch of great sounds. All completely free.

Browse and download them HERE.

Our Favourite: Recording Hybrid Kit

Boxed Ear Free Kit Packs

Although Boxed Ear doesn't boast as gigantic library, what they do offer is amazing.

Their kits sample some lesser known machines like the Roland CompuRhythm CR-78. They even provide a preview of the sounds before you download.

Grip them for free HERE.

Our Favourite: The Dave Smith Instruments Tempest Vs. Akai MPC60

Goldbaby 'Free Stuff' Section

This site is packed with valuable sample kits. There's 19 free packs in total. They even have a sample pack of the first ever drum machine! The Wurlitzer Side Man.

Make them all yours right HERE.

Our Favourite: The Cassette808

Rhythm Lab Sample Bundles

What sample collection would be complete without the Amen Break? Rhythm Lab offers it and more in their lengthy free sample library.

Just be sure to click the FREE button on the right side of the page.

Get them all HERE.

Our Favourite: Yamaha CS15 Loops

Happy Producing

Now that you've fattened up your production tools don't forget to use them. Always be producing.

And let us know your favourite free sample packs. We're always looking for new samples too! Because no toolbox is too big.

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8 Free Sample Packs Every Producer Needs | LANDR Blog

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Sample packs are essential for any frugal producer. Especially when they're free. They capture all the analogue goodness without the price tag. No choosing between buying food or saving up for that TR-808. Free sample packs make you a better producer-because a musician is only as good as their tools right?

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You don't have to buy expensive royalty-free sample packs for making music when there are plenty available for free download on the internet. Scott Wilson picks 21 of the best to give you fresh inspiration.
Whether you agree with the ethics of sampling or not, modern electronic music was largely built on the practice. Digging for old records and finding the perfect loop to cut, pitch and stretch is still one of the best ways to find inspiration, but there's no shame in using an official, royalty-free sample pack to make music.

Licensed packs can be purchased from stores such as Beatport, Juno, Irrupt, Sample Magic and Loopmasters, but it's possible to find great, royalty-free sample collections for no money across the internet. These range from professional packs released by stores as promotional freebies to ZIP files made at home by artists and synth enthusiasts.

To make the process a little easier, we've collected 21 of our favorite free sample packs from across the web. Whether you're using a hardware MPC sampler or you just want to play around with Ableton Live's Simpler or Drum Rack instruments, there's enough variety in the packs below to make hip-hop, techno, house, ambient, experimental club music and more.
Legowelt's Yamaha DX Files
Danny Wolfers aka Legowelt's biggest pack of samples covers Yamaha's family of FM synths, including not just the DX7 used on everything from A-ha's 'Take On Me' but the DX100 beloved of Detroit techno producers like Derrick May and Robert Hood. Perfect for adding a touch of '80s class.
Legowelt's Korg Mono/Poly samples
Another product of Danny Wolfers' studio, this free sound pack comes from the underrated Korg Mono/Poly synth, delivering everything from "juicy big basses" and "occult space pads inter-dimensional arpeggios" to "futuristic synthetic drums" and "metaphysical atmospheres".
Mega Drum Machine Collection
This giant sample pack contains all the beats from a staggering 150 drum machines, from the famous (Roland's TR-808, Korg's Minipops) to the not so well known (Cheetah's SpecDrum, BME's Rattlesnake and Sakata's DPM48). It's mapped for NI's Kontakt, but if you don't have that you can just take the individual sample files and use them however you like.

The Nintendo Game Boy's crunchy, 8-bit drums aren't just for chiptune artists - they're a useful texture to have no matter what music you make. Add some distortion or compression to beef them up or put them into a sampler and you've got a gritty sound that'll work in everything from techno to ambient music. One of several great packs from the BPB site.

One of the best packs from Wavy's portal of free user-created sample packs, Beached gives you a small but useful selection of dub techno-style synth sounds that'll also find work for ambient productions.

This huge collection from The Pro Audio Files could end up being the only drum sample pack you ever use. Covering everything from primitive synth rhythms to acoustic drum loops, it also features a number of unique one-shot samples that'll add character to any Ableton Live drum kit.
QuBit Electronix modular jam
If you want to add some modular synth bleeps to your music but don't have the cash, this massive pack of one-shots, loops and stems from Eurorack manufacturer QuBit Electronix is highly recommended. One of the more unusual offerings from Converse's giant Rubber Tracks library, a huge resource of royalty-free samples worth checking out.

Another gem from the Rubber Tracks site, this pack of synth samples from Minimal Wave legend Das Ding will give your music some vintage analog grit, covering everything from ambient loops to lo-fi kicks.
Irrupt's Jam Sandwich Vol. 1
If you just need a basic, meat and potatoes sample pack for making big room dance tracks, this free collection from online store Irrupt has all you need. There's 33 bassline loops, 60 melodic loops, 51 drum loops and 22 vocal hooks, a solid addition to any library.
Buchla Music Easel drums
Don Buchla's cult Music Easel synth is still prized by open-minded musicians, but they're not easy to get hold of. This collection of abstract, alien drum sounds is the perfect alternative for anyone with 808 and 909 fatigue.
TĀLĀ's Duchess No.1 sample pack
After making a track in 10 minutes for FACT TV's Against The Clock series, TĀLĀ gave away her own personal pack of samples used to produce it. Expect bold kicks, Thai drums, sitar riffs and a "Serbia shaker".
SampleRadar ad-lib vocal samples
SampleRadar's exhaustive collection of free samples is basically a database of those sounds you'd get on music production magazine cover CDs. This one gives you a relative rarity in the world of free samples, a collection of royalty-free vocals to chop and pitch how you want.
Mike Huckaby's Groove sample pack
Detroit veteran Mike Huckaby gave away this sample pack via Groove magazine back in 2012, containing live percussion, synths recorded with his beloved Waldorf Wave synthesizer and samples from the classic Korg Triton.
RE:VIVE's Madagascar, Switzerland, Tahiti, China and an accordion pack
RE:VIVE's speciality is in collecting sample packs of openly licensed archival material. This set is as unique as it gets, featuring field recordings and performances from Madagascar, Tahitian pop sourced from old 78s, Swiss Alpine music and Chinese accordion marches.
Samplephonics Abstract Percussion
Samplephonics is another hub for paid samples, but it's got plenty of free content to download as well. Abstract Percussion is one of the most unusual, featuring a set of obscure percussion ideal for crafting your own fourth world music.
Commodore 64 synthesizer sessions
Another great pack from the BPB site, this will give you a collection of 8-bit synth sounds from the Commodore 64 computer from the '80s. Perfect for making a retro video game soundtrack or just adding texture to a hip-hop beat.
Filter Dread sample pack
Filter Dread's hybrid of jungle, rave, grime and ambient music has made him one of the UK underground's most unpredictable artists. This pack gives you all the oddball drums, synths and zapping effects to to make your own zoned-out bangers.
Samplephonics Live Jungle Breaks
Sometimes you just want a pack of drum breaks to make some jungle, and this free pack from Samplephonics is ideal. If nothing else, it makes a change from hearing the same old amen break all the time.
NASA space and shuttle samples
If you want a create a build-up from the sound of a space shuttle taking off or just need something really loud for a noise track, NASA's sound archive has you covered. The various beeps and radio sounds are great for creating ambience too.
Philharmonia Orchestra samples
Orchestral samples generally very expensive to buy, but the UK's highly regarded Philharmonia Orchestra has a gigantic royalty-free database for download at its website. It's not just string sections either - it covers everything from the French horn to the Thai gong.
SampleRadar's free jazz club samples
Musicians have been sampling old jazz records since the MPC was invented, but this SampleRadar pack won't get you in trouble. It's got live bass, electric piano and dusty drums that will work in deep house and hip-hop beats.
Read next: Free music production software: 10 VSTs and DAWs for making great tracks with no money

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21 great free sample packs for producers on a budget

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You don't have to buy expensive royalty-free sample packs for making music when there are plenty available for free download on the internet. Scott Wilson picks 21 of the best to give you fresh inspiration. Whether you agree with the ethics of sampling or not, modern electronic music was largely built on the practice.

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Drum Kits 12 Piece Pearl Master Custom Drum Kit Basement Drums Big Mono Brush Kit DrumTasteFree Easyrider Free EXS24 Drumkit and Samples Free Jazz Funk

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In this step by step guide to using Logic's updated delay plug-ins Mark Cousins takes a look at stereo and dub delay...
Stereo Delay

7: Now let's explore the application of Stereo Delay on the second synth track. The Stereo Delay has a different delay line for each respective channel (left and right). In this example, we've loaded two different dotted settings on each channel.

8: The Routing button lets you quickly explore a variety of signal routing options between the two channels of delay. The Ping Pong setting, for example, creates an effect where the delayed signal is passed from one channel to the next.

9: Some of the more complex routing options include both Crossfeed and Feedback setting, as well as phase reversal. The result is a more complicated delay, which appears to move around the stereo sound stage and surround the listener.

10: The Stereo Delay doesn't have the distortion options included on the Tape Delay, but it does have Low and High Cut filters on a per-channel basis. In this example, we've reduced the High Cut on both channels, and the Low Cut on the left channel.

11: Sometimes, the extremities of the Stereo Delay can appear to be too wide for a mix. In these situations, try inserting an instance of the Direction Mixer after the Stereo Delay. Reducing the Spread control will make the delays progressively monaural.

12: If you want the delays to recede in the mix, consider placing an instance of Space Designer after the Stereo Delay plug-in. The plug-in will default to a 100%-wet setting, though, so you'll need to balance the Rev and Dry faders accordingly.

Dub Delay

13: In this last example we're going to explore the application of dub-like delays using the second Logic Project. Start by creating a duplicate track lane, with a copy of the drums region dragged across, set to No Output in the mixer.

14: On the new track, edit the drum loop so that the region is only active when you want the dub delay effect to occur. You could do this by automating a bus send, but the 'edited region' solution is arguably easier to visualise.

15: Now create a bus send from the track with the edited region, with an instance of Tape Delay. The setting uses a 1/16 dotted setting with plenty of high-end roll-off, a lowered Clip Threshold and a touch of Flutter to degrade the sounds.

16: For another level of interest, we've added an instance of Tremolo after the Tape Delay plug-in. Using the Slow Panning setting, the dub delays appears to swing from side to side in the mix. Lower the Depth to lessen the width.

17: As well as switching the effect in and out, dub is often characterised by variations in the amount of feedback. To start automating the Tape Delay plug-in, use the menu function Options > Create Tracks for Selected Channel Strips.

18: Enable the Automation View (A) and select the Tape Delay: Feedback parameter from the drop-down menu in the track list. Now draw in some automation moves, increasing the level of some of the points up to 100%, while lowering others.
This tutorial is endorsed by Point Blank. With courses in London, online and now in LA, Point Blank is the Global Music School. You can study sound to picture on their Music Production Diploma courses, with pro industry tutors.

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Logic Pro X Tutorial: Logic's Updated Delay Plug-ins - Step-by-Step

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In this step by step guide to using Logic's updated delay plug-ins Mark Cousins takes a look at stereo and dub delay... Stereo Delay 7: Now let's explore the application of Stereo Delay on the second synth track. The Stereo Delay has a different delay line for each respective channel (left and right).

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Getting your head around the different types of compressor can make all the difference to your sound. Mike Hillier finds out what's involved...

Knowing how to set your compressor's threshold, ratio, attack and release may be the biggest part of the battle, but knowing which compressor to use for which job in the first place can help to make the most out of the dynamics processors in your collection.

By picking the right compressor, you can achieve a certain amount of colour and EQ without having to reach for any other tool. Sometimes simply running your signal through a certain compressor can produce the effect you require without any gain reduction at all.

The first thing to look for when picking a compressor is what type of gain reduction technique it uses. VCA (voltage controlled amplifiers) compressors are probably the most common hardware design, and there are plenty of software emulations.

This technique is usually chosen because it offers a wide range of potential, from very fast to very slow, and can be configured to respond with a very clean, transparent sound. The basic operating principle is that the signal passes through a detector, which converts the AC sound voltage into a simple DC voltage. This DC voltage is then used to control the gain of the amplifier.

Optical compressors use an ingenious technique to control the amount of gain reduction. The audio signal is sent through a lightbulb coupled to a photocell (a light-dependent resistor). The louder the signal, the brighter the bulb, this reduces the resistance of the photocell causing greater gain reduction. The bulb and photocell have a built-in lag time, which gives optical compressors a smooth but slow compression quality.

The gain stages of an optical compressor can be either valve or solid-state, with some older models, such as the LA-2A, having a coloured sound compared to newer models such as the Maselec MLA-2.

The last of the 'big 3' compressors is the FET compressor. This uses a Field-Effect Transistor (FET) to vary the gain. Initially FETs were chosen to replicate the behaviour of valves, but, as anyone who has heard an FET compressor will know, the two have very different characters.

FET compressors, such as the 1176, are incredibly fast and able to produce a considerable amount of gain reduction without sounding forced. They can be great for pushing sounds forward in a mix, or for slamming transient material heavily.

Some of the oldest compressors use valves to vary the gain and these compressors, such as the Fairchild 670, are commonly referred to as vari-mu compressors. This class of compressor includes many of the most revered compressors in existence. Vari-mu compressors are quite sluggish compared to more modern designs. While this makes them fairly useless at squashing transients, it can make them wonderful at bringing up the body of a sound while leaving the transient untouched, so they are often found on drum busses, mix busses and vocal chains.

When the BBC, EMI and so on first tried to replace valve compressors, they used diode-bridges as the compression stage. These are rare, but can still be found in designs such as the Neve 33609. The problem with this is that the signal has to be reduced significantly before it can be pushed through the diode, meaning a considerable amount of gain is needed on the output, resulting in quite a high noise floor.

An often overlooked compressor type is the PWM compressor. This turns the signal on and off at rates far higher than the audio range to reduce the level.

If the signal is on and off equally, then it is reduced by 50%, whereas if it is only off for a tenth of the signal, the energy is reduced by only 10%. This difference is called the pulse width. PWM compressors built with modern components can produce very clean compression with almost no artefacts.
Crane Song and Great River both produce PWM compressors.

Many modern digital compressors can model multiple types of compression in a single plug-in, without emulating any single specific compressor model.

The FabFilter Pro-C (pictured here) has three modes: Clean, Classic and Opto. Compressors such as this often have variable knee and many other features that aren't present in hardware emulations, making them a great choice for almost any channel in your mix.

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Pro Tools Tutorial: Compressor Types Explained - MusicTech

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Getting your head around the different types of compressor can make all the difference to your sound. Mike Hillier finds out what's involved... Knowing how to set your compressor's threshold, ratio, attack and release may be the biggest part of the battle, but knowing which compressor to use for which job in the first place can help to make the most out of the dynamics processors in your collection.