For some of us this may be a very mysterious image: “panning” presets in a delay device of a sequencer DAW…

Well it IS a mysterious image until you check this article about the Haas effect.

Have you ever been so frustrated with a mix that doesn’t have life, wide stereo image and airy, “natural” space? Are you not getting enough depth in your mix? Have you added so much stereo reverb to add “space” to your mix that you end hating what you have done? Well… I have…

It is truly horrible getting every track in a mix sounding so “mono” after panning left and right different lines or instruments. It feels truly like there is something in the way from the original tracking to your final mix. The panning of some tracks helps very much to make a clear mix but sometimes that is just not enough to make things clear and certainly deeper, spacious, open and rich-sounding.

There is a solution to this lack of depth: The Haas Effect . It can take a simple mono instrumental or vocal line and give it presence or take it to the back of the mix, depending on how you use it and what other effects you add to the chain.

Basically, what you are doing with the Haas effect is making the listener’s brain to interpret the sound coming from a certain direction and angle in a way that is more natural to the ear than a simple panning adjustment. The Haas effect takes full advantage on the fact that we have TWO ears.

In real life, when a sound comes from the left it is received by the left ear BEFORE the right ear, so the brain interprets this difference as “a sound coming from the left”. The interpretation depends on how long is the delay between the two ears the less the delay, the more centered is the sound. This very short delay can be interpreted as a phase shift because the sound reaches first one ear and then the other. So, when there is no phase shift, no delay between the two perceived signals together are interpreted as dead center, with no panning at all.

We CAN use this psychoacoustic effect (the Haas effect) to print more depth and directionality into our mixes without even moving the panning control of the console! That’s why I put the snapshots from Ableton’s Simple Delay. You can make your own Haas effect presets in any delay that permits very, really shot delay times and also inserting a time shift between the two channels. The Haas effect is performed delaying one of the two channels (left or right) just a little bit, from a couple of samples to no more than 30 milliseconds (more in depth scientific info in the article from Wikipedia referenced above); the channel that reaches our ear first is the one that dictates where is the sound “coming from”, the later channel is interpreted as the natural tail of the sound.

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We love Mixing (at least I do). Mixing is always challenging and fun, it’s a stage where you can take some creative licenses (that you didn’t or couldn’t when composing) in the type of sound and expressive response you want to print in the final representation of the work. You become the director of the project, while you still have to take care of not ruining the composition and it’s first, original intention. It is truly a universe of its own in audio creation.

In my experience I have to say that when anyone begins the process of mixing for themselves or for someone else, they do it (and I did it too) thinking about mixing as a simple volume and panning tweak. Wrong!

Both volume and panning are the very main pillars of mixing since stereo audio was possible and since traditional analogue mix consoles were invented, but, as we all know, there are dozens of other factors that get into the picture and change it when you are trying to take your program into the next level.

One of these factors is Monitor Gain and its relationship with the way you may mix your records in the analogue console or DAW faders.

A very frequent mistake for every beginner in mixing (a mistake I committed a long time ago) is to try to “balance” the volume of the summing buss so it stays close to 0dB. I think this happens because we try to get the “finished-CD” sound quality out of the box without thinking about mastering.

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Have you got a knack for writing and knob twiddling as well? I’m looking for talented writers and electronic music producers to write for this blog. Please reply to this thread or to modcam@gmail.com if you’re interested in writing articles/tips for emusictips.com!

I’ve gotten a few requests to make the top 10 signs your track is amateur for electronic music instead of acoustic music. Well, here’s my list of things you should learn to avoid if you want professional sounding tracks.

1. As I’ve written about before, the most common thing that prevents amateurs from getting a full sound is not filling the “box” that is volume, panning, and frequency. The typical dilemma is this: as more sounds are layered together, the audio may start to clip. And so, you turn the gain down on the each channel of the mixer so it doesn’t clip. But then, it sounds quiet. In order to fix this, you need to learn about compression and mixing. If used properly, compression reduces the variations between one audio channel’s highest and lowest gain levels throughout the track, which allows you to turn the volume up without clipping.
2. Muddy sound:

   When too many frequencies are overlapping in a mix, the result is “muddy”. To prevent mud, you must consciously keep in mind what range of frequencies you are adding with each new part. Inevitably, frequencies will overlap, no matter what instruments you choose. For example, two bassy sounds on top of each other will interfere, resulting in weird phasing issues. If you want to use two instruments that use up the same frequency spectrum, you’ll want to carve out the highs on one and carve out the lows on the other (through the use of EQ, you will eliminate too many overlapping frequencies and clear up your mix) The end result should be consist of many different parts that all cover different ranges of frequencies, which all add up to a full, clear sound.

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download “10 ways to get your music into film and TV” (60kb PDF)

via www.filmmusicmag.com

There is one basic fact about the film and television music industry that drives much of what you will read in this guide: it is a very, very competitive business and there are many more songs and instrumental music pieces than there are openings and places to use them in film and television. In Los Angeles on any given day, hundreds, maybe thousands of people are marketing their music for film and television productions. This guide is designed to show you how you can successfully compete in this industry, whether you live in Los Angeles, New York, or in a small country town far removed from the major music cities.

“Location, Location, Location!” – The tried but true real estate mantra is definitely applicable to the film and television music business. A simple fact: being in LA or NY can make it easier to compete for work. While film and television shooting locations can be found worldwide, the infrastructure for post production, which includes music, is still centered in Los Angeles. Although this is changing rapidly as cheap digital editing equipment becomes available in other cities, in film work, the city that the director resides in can also be a major factor in underscore work.

It’s useful to note that song placement is much less location-oriented than score composing. Score composing requires a weeks-long cycle where it can be very helpful if the director and composer are in close physical proximity so demos can be heard. Song placement is much more easily done from locations outside of LA since once the director or music supervisor decide they want to use a song, the physical location of the songwriter is not that important.

That much being said, if you’re in LA or New York, make the most of it and seek out personal relationships with people in the business. Film directors, television producers, and music supervisors are among the most important people you can meet in terms of getting your music into film and television projects. By putting a “face with a name,” you can increase the chances of your music being heard.
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via HomeTracked.com

I keep a collection of audio samples designed to help check my monitor setup. Test tones, essentially, that I use after I’ve moved my speakers or desk, to ensure the speakers still behave as they should.

I’ve included 4 of the samples below, and I hope you find them useful - and possibly enlightening. Each tests a facet of the two most common monitoring problems in home studios: Uneven bass response, and poor stereo imaging.

Sine wave sweep

Contents: A sine wave sweeping from 40Hz to 300Hz.
Use this to test for: Bass response, sympathetic vibrations.

Unless you’re outdoors, or listening on headphones, you’ll notice the volume rising and falling as the audio plays. That’s normal, although the level doesn’t actually change. (Open the MP3 in your DAW to confirm this.) Rather, you’re exposing the acoustic response of your room.

Use this test as a rough gauge of how extreme the acoustic issues are in your space. (You can flatten the response somewhat, but acoustic treatment is a topic unto itself. For some more information, check the quick backgrounder on home studio acoustics.)

Additionally, the sweep can expose low-frequency dependent rattles, buzzes, or other sympathetic vibrations happening in the area around you. With this test, I once discovered the casing on an overhead light shook at exactly 140Hz, after puzzling with a mix for 15 minutes, unable to isolate the odd rattling sound.

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Download the Ableton Live 7 techno beat project (.zip format) to play around with this example track

In the techno beat live set provided above, you’ll find several samples:

• kick.aif: this is a clean, punchy kick that works well for techno. It has a nice click at the beginning, followed by a punchy sine wave that bends in pitch after a few milliseconds
• snare.aif: this snare has punch and smack to it, with plenty of high-frequency content
• open hh.aif: a very short, minimal sounding click that works nicely for anything minimal
• boink.aif: a high hat and a cup type of sound.
• blip.aif: a synthesized little hit with high and mid frequency content
• tom.aif: a clicky, synthesized tom, very techno

These samples came from SampleMagic’s Minimal/Tech House sample and loop library.

Here’s a quick tutorial of how I made this beat:

1. Create a new Drum Rack Instrument by dragging it from “Live Devices” into the clip/device drop area
2. You should see a grid of 16 squares in the new Drum Rack. Navigate your samples directory with one of the File Browsers, and drag your drum samples into the grid squares. Note: you can select multiple samples and drag them into the drum rack at once, they will be automatically placed in order.
3. If you want your drums to be velocity-sensitive, (which is a must for writing expressive drum paterns) then you will want to select each drum except the kick (either by clicking on a square or selecting it in the Chain List) and set Vel (on the right side) to 50%. The reason I don’t like to set the Vel on the kick drum is because the kick is your constant, you may not want to accidently make the volume of one note louder than another.
4. Now you’ve got a nice drum kit to work with. The typical house or techno beat has a 4/4 kick drum, so go ahead now and create a clip by double clicking an empty slot in the drum rack track, and then double clicking on the newly created clip.
5. Before you start creating any notes, make sure your clip has its Groove set to “Swing 16″, and the global groove amount set to anywhere from 20 or so to 70 or 80. This will make your clip play back with a lot more swing, and this is essential for creating a “groove” feeling. See the image below for where to find the global groove amount:
   
6. Create a Kick drum note on 1, 1.2, 1.3, and 1.4. Play the track so far by pressing space bar, and listen to your simple beat. Now click on the clip and hit Cmd+D or Ctrl+D for PC users. This will duplicate the clip.
7. Now let’s add in some snare in this new clip. on 1.2 and 1.4, add a snare drum note. This is the simplest of all house beats.
   
8. With each new duplicated clip, begin to add in more samples in between the kicks, making it however simple or complex as you’d like. By adding new clips to the drum rack for each new variation, you make it easy to create a progression that will fit into a track. It should start out simple and grow more complex as the song goes on.
   
9. Now that you have several different clips of different beats, you can click record and then trigger each clip and let it play for 2 bars or so, then click the next, then the next after that. This will create a recording of your beats, which you can view by pressing Tab to switch to the song mode.
10. That’s it for now. Good luck! Oh, and if you have any of your own tips, post them here in the comments.

Related articles

• Sonic Transfer: How to build a kick drum
• Sound On Sound: How to synthesize a snare drum (advanced tutorial)
• Tweakheadz: How to write original drum tracks

1. Jailbreak your iPhone with ZiPhone (download for Windows and Mac)
2. Download Masayuki Akamatsu’s akaRemote (download .zip here) and upload it to your applications folder with Fugu (Alternatively, use the Installer app that comes with ZiPhone, and find akaRemote under the “Network” category)
3. Install i3L Midi Bridge software for Mac OS X (download .dmg here) This will allow you to intercept the controller data being sent from the iPhone
4. Open akaRemote on your iPhone, and go to the configuration page. Double tap the series of numbers after “Host Address”. Change this to the ip address of the computer that will receive the MIDI data. On most home wireless networks, it will be something like 192.168.0.X, or 192.168.2.X on other routers. I don’t know if this will work without a wireless network router.
5. Open i3L and then run akaRemote on your iPhone. When you press buttons and move sliders in akaRemote, you should see corresponding activity in i3L’s buttons and sliders.
6. In Ableton Live’s preferences, under “MIDI Sync”, you should see “Input: from i3L_v0.2 1″. Turn on Track and Remote.
7. In Live, hit Command + M (Ctrl + M for PC users) to access the Midi Map mode. Now click any button or slider that is highlighted in blue. Now press a button in akaRemote on your iPhone. If everything is correctly configured, you should see a number appear in a box over the blue highlight. This is the MIDI CC number of whatever button you pushed.
8. Hit Command + M again to exit Midi Map mode. Now press the buttons that you have assigned and they should activate the ableton live buttons.
9. Awesome!

These days, I’ve been hearing a lot of criticism aimed at the compact disc format. Vinyl is regaining popularity as people are realizing that CD’s just don’t sound the same as vinyl. Now, whether this is just a placebo effect, I’m not sure. Apparently, audiophiles can’t tell the difference between Monster Cable and coat hangers. Should we trust the human ear so much to say that we can really hear the difference between CD and Vinyl? The differences are there, surely. Vinyl carves a smooth, continuous groove around the disc, whereas CD reduces the audio quality to 44,100 slices, each having 65,536 possible levels. I’m willing to bet your average listener couldn’t tell the difference. Another thing about vinyl is that as the record progresses, more and more high frequencies are lost.

“Most people don’t realize that the distance around the inside of a 12-inch record is about half the distance than around the outside,” Golden explains. “As the distance around each revolution decreases, the high frequencies become harder for a playback stylus to read.”
Link to source

On a tangentially related side-note, I found an interesting video of how vinyl records made, check out How Vinyl Records Are Made Part 1 and Part 2. I know the vinyl vs. CD format war will never be resolved, but it’s interesting to consider when deciding between the two methods of physical distribution for your project.

In this short video, I demonstrate how to load an audio file into the Simpler instrument to create a monophonic “human flute” sound. This technique can be applied whenever you need to create a playable instrument from a single recorded tone.

View movie

If you aren’t confident enough to record melodies from a MIDI keyboard or even your computer’s keyboard (a nice feature in Live for when you’re on the road with no MIDI controller), I find that the easiest way to write melodies with the pencil tool (Command + B for mac users, Ctrl + B for windows users) is to write in your melodies step by step. If you recall the formulas for major in tones (W = whole step, H = half step) (W W H W W W H) and minor (W H W W H W W), then you can use the Fold feature of Live’s clip view to hide the notes that are not included by one of these formulas. Notice in the first image, we have one octave of notes stacked up on top of each other in two different scales, F major and F minor.

All we need to do is create one of these stacks in a MIDI clip, and then duplicate it once or twice. Just select all the notes, then hold down option (mac) while dragging the notes up one octave. This should create a duplicate of your notes, but transposed up one octave. Do this again for the octave below. Now when you click the “Fold” button located at the top left of the clip view, all notes that are not in the clip are hidden. Note in this second image that at the very left, there is a stack of notes that form the scale of F minor. After that, I randomly double clicked to create new notes all over the grid. I set my synthesizer’s polyphony to 1 so that it can only play one note at a time. So no matter what notes I drew, they were all in key. As long as you have the fold view enabled, you can now draw notes anywhere and it will still sound pretty decent.

by Bob Katz

Mastering requires an entirely different “head” than mixing. I once had an assistant who was a great mix engineer and who wanted to get into mastering. So I left her alone to equalize a rock album. After three hours, she was still working on the snare drum, which didn’t have enough “crack”! But as soon as I walked into the room, I could hear something was wrong with the vocal. Which brings us to the first principle of mastering: Every action effects everything. Even touching the low bass affects the perception of the extreme highs.

Mastering is the art of compromise; knowing what’s possible and impossible, and making decisions about what’s most import and in the music. When you work on the bass drum, you’ll affect the bass for sure, sometimes for the better, sometimes worse. If the bass drum is light, you may be able to fix it by “getting under the bass” at somewhere under 60 Hz, with careful, selective equalization. You may be able to counteract a problem in the bass instrument by dipping around 80, 90, 100; but this can affect the low end of the vocal or the piano or the guitar - be on the lookout for such interactions. Sometimes you can’t tell if a problem can be fixed until you try; don’t promise your client miracles. Experience is the best teacher.

Think Holistically

Before mastering, listen carefully to the performance, the message of the music. In many music genres, the vocal message is the most important. In other styles, it’s the rhythm, in some it’s intended distortion, and so on. With rhythmic music, ask yourself, “what can I do to make this music more exciting?” With ballads, ask “is this music about intimacy, space, depth emotion, charisma, or all of the above”? Ask, “How can I help this music to communicate better to the audience?” Always start by learning the emotion and the message of the client’s music/ After that, you can break it down into details such as the high frequencies, or the low frequencies, but relate your decisions to the intended message of the music. Some clients send a “pseudo-mastered” demonstration CD illustrating their goals. Evin if you don’t like the sound on their reference, or you think you can do better, carefully study the virtues of what they’ve been listening to. During your mastering, refer back to the original mix; make sure you haven’t “fixed” what wasn’t broken in the first place. There is no “one-size-fits-all” setting, and each song should be approached from scratch. In other words, when switching to a new song, bypass all processors, and listen to the new song in its naked glory to confirm it needs to be taken in the same or different direction than the previous number. Likewise, as you gain experience, you may want to “tweak” the “presets” in your equipment. Presets are designed to make suggestions and provide good starting points, but they are not one-size-fits-all and should be adjusted according to the program material and your personal taste.

To continue reading, download the PDF for Secret of the Mastering Engineer

Hey everybody, sorry for the lack of posts lately. I’ve been focusing my efforts on my own musical projects. But good news, my debut album has been unleashed into the world, and I should have more time to keep updating the site.

My album, Six Minute City, is available for listening and for purchasing online at modcam.com. Additionally, you can download free tracks from the album on last.fm

Here at emusictips, I’m always on the lookout for fresh new electronic sounds. Artists such as Shulman, Bluetech (Evan Bartholomew), Kilowatts, Pitch Black, and Shen have piqued my interest because of the technical mastery evident in their sound. Here’s a short list of the things I think make their music great:

Conscious use of space: just like any good graphic designer will tell you, space is important. In design, space comes in the form of white space, which is one of the most important elements in creating aesthetic compositions. The same thing applies for music. Allow your listeners to breathe, so to speak. You give them space and they will appreciate it.

Conscious use of effects: One of my favorite things to add to any synth is a series of effects and processors that polish the sound and make it pop out of the track. Delays are great for filling in empty space that you’ve created between elements in the track. Try adding a 3/16th delay to any sound and then adjust the feedback to your liking. This will create a sound that repeats every third sixteenth note, and will gradually fade out. But do not overdo it! If you have a lot of feedback, only play a note every so often so that you can still retain that space that is so important. Also, if you’re going to add effects such as phasers, flangers, distortion, etc., make sure that not all of the instruments in your track are layered with these kinds of effects. The purpose should be to make a particular sound stand out from the rest to create contrast.

Automation: To keep me interested as a listener, you need to develop movement in your song. Movement requires changes along time. The best way to achieve this is to automate knobs and sliders in your software sequencer. When you’re tweaking knobs on a synth or sampler and you find that turning a certain knob sounds cool, hit record and then record those knob movements in real time. Go back over the song and repeat the process as necessary to create a multi-faceted track with lots of movement.

I was recently contacted about some sample web sites and decided to put a little review of the best ones here.

SoundSnap.com has a nice collection of sounds categorized by large icons.
Pros:

1. Built-in audio player with a picture of the waveform makes browsing through sounds quick and easy.
2. No signup required
3. User-contributed sounds
4. Tag cloud with popular categories of sounds
5. All samples free for use under license similar to Creative Commons
6. MP3 and WAV formats

Cons:

1. None that I can point out so far

The Freesound Project has been a personal favorite of mine for a while now. It’s got loads of sounds, from high quality field recordings to synthesized ambient soundscapes, they cover the spectrum of sounds, and best of all, they’re all Creative Commons licensed.

Pros:

1. Built-in audio player with a picture of the waveform makes browsing through sounds quick and easy.
2. Geotagging capability
3. User-contributed sounds
4. Remix tree shows sounds that have been remixed by other users
5. All samples free for use under Creative Commons
6. Sample Packs offered for quick acquisition of multiple sounds

Cons:

1. Free signup required
2. Design is a little dated

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I just found this helpful tutorial from The Whipping Post

And here is a basic overview of the tips:
1: Use MONO Sound Sources
2: Rest your ears
3: Keep the bass and kick panned dead center
4: Use EQ to cut, not boost
5: Fix frequency masking problems
6: EQ boosting also boosts your volume — keep this in mind when setting relative levels
7: Subtle effects are most effective — contrast is key
8: Use noise gates strategically, and before reverb
9: Cut off unnecessary frequencies, especially low rumble below 30-40hz
10: Avoid mixing with headphones

I found a helpful article that will teach you some fundamentals about audio. If you have ever wondered what a sound wave actually is, or what’s behind all the complex audio terminology, take a few moments to read this article and fill yourself in on the background knowledge that will give you the upper hand as a producer.

Here’s what’s covered in the article:

* what sound is caused by
* what a sound wave is
* what a cycle is
* what frequency is
* what a hertz is
* what a pure tone is
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from hometracked.com

We’ve all experienced it: 3 seconds into a track you’ve never heard, you know instinctively that it was recorded and mixed in someone’s bedroom.

Amateur recordings often sound “amateur.” But what differentiates these hometracked opuses from professional recordings? It’s not just fidelity or sonic quality: Many competent engineers produce lo-fi or distorted mixes on purpose, when it suits the song. Rather, amateur recordings tend to share some key traits, telltale signs that the mixing and recording are the work of a novice.

You can learn to recognize and address these traits in your own recordings, and produce more polished, professional mixes:

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adapted from http://www.kubton.com/eq.html

I have read comments that compression is the most misunderstood audio process but I really think it is equalization. It might be the most over and under used audio processing tool at the same time. What complicates the matter is factors intrinsic to our humanity. We perceive different frequencies different ways. Some frequencies will sound louder or quieter than their actual volume. This is why some audio hardware has a “loudness” button. Most people will try to improve sounds by boosting frequencies. But EQ is a sculpting process. The best result will be attained by boosting and cutting. You can accentuate one frequency by reducing another frequency , and it not make the sound muddy. Following is a basic overview of the spectrum of audible frequencies.

20-40hz: Edge of human range of unwanted rumble often complete removed.
40-80hz: Sub-bass or “feel” of bass. Can add low end kick or over power mix. Is not produced by small speakers..
80-250hz: Bass 100-200hz can be boosted to add fullness or cut to reduce boomy sounds.
250-600hz: Fullness or some vocals and percussion. The cardboard box sound of kick drum is around 300-400hz.
600-4khz: Midrange all too easy to add mud. 800hz is where the “cheap” sound comes from. 2k-4khz is where the attack of most percussion and some other instruments reside.
4-6khz: The “presence” range that determines how far out in front of the mix vocals sound. Can easily become grating.
7khz: The nasty realm of sibilance, the unwanted “s” hiss
8-20khz: This is the range of “air” or “brilliance”, and its presence adds sparkle.
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Granular synthesis is the creation of sound by taking tiny slices of audio from another source and playing them back really quickly. Here’s a handy trick for chopping up samples into pieces without ever leaving Reason. First, create a Redrum drum computer. Next, notice that the function of the bottom left knob in the Redrum varies depending on the channel number. In the third, fourth, and fifth channels, the function is “START”. What this does is alter the starting point in the sound file that you have loaded. So, load up any sound file of your choice (vocal sounds are good) into channel 3, 4 or 5. Then turn Channel 3’s length knob to the left. You want the sound to be just a short burst. Next, while holding Shift, right click (or command click for mac users) on an empty space in the rack. Select “Matrix Pattern Sequencer”. Now, flip the rack around with the Tab key, and then drag a cable from “Gate CV” on the Matrix to “Gate In” on the Redrum. Flip back around again, and then press the Run button on the Matrix. You shuld now hear a stuttering sound. Now play with the Start knob and you will hear the effect we were looking for. Now you can record the automation of the start knob (make sure the Record button next to “Redrum 1″ on the sequencer is lit up red, hit record and then press play). Go wild with the pitch and the start time, and you should get some cool sounds. Even cooler: play with the Resolution knob in the matrix to adjust the speed of gate triggering.

Download the Reason .RNS

adapted from the iZotope Ozone guide to mastering.

UPDATE: This article is merely a brief introduction. Here are a few links to check out for more detailed information:

• TweakHeadz Guide to Mastering in the Home Studio
• WikiPedia entry on Audio Mastering

Mastering most commonly means the final step in the recording process, before the music goes off to be pressed into CD or vinyl format. It is a process that involves creating consistency among all of the tracks on an album so that they fit cohesively. This process is typically carried out by skilled audio engineers who have very trained ears. All record companies employ professionals who master all material, even if the mixing is spot on.

It is important to note that mixing and mastering are worlds apart. Recording and mixing all occur within the sequencer of choice, but mastering is applied to an audio file that is already mixed down, outside of the sequencer. My personal favorite choice of mastering tools is izotope ozone, because they offer the tools you will need: equalizer, reverb, multiband compression, volume maximization, stereo imaging, and harmonic excitation.

More and more today, bedroom producers are emerging, and they are taking on many roles, including songwriters, producers, recording engineers, and they are even mastering their own material. When you are on a tight budget, it’s impossible to afford the high prices of mastering services, so many of you would like to learn to master on your own. However, mastering is best when it is done by someone other than the producer, because you as a producer are too close to your own music. You won’t notice things that another pair of ears will notice, and therefore you may improperly master your own material without realizing it. If it’s possible, always get someone else to master your music. That being said, if you’re not going to get your music professionally mastered, at least learn how to mix and apply mastering effects to get the best sound possible.

First, what’s wrong with my song?

• It’s not loud enough. It sounds wimpy next to other CDs. Turning it up or mixing down at a higher level doesn’t solve the problem. It sounds louder, but not, well LOUDER.
• It sounds dull. Other CDs have a sparkle that cuts through with excitement. You try boosting the EQ at high frequencies, but now your song just sounds harsh and noisy.
• The instruments and vocals sound thin. Commercial songs have a fullness that you know comes from some sort of compression. So you patch in a compressor and turn some controls. Now the whole mix sounds squashed. The vocal might sound fuller, but the cymbals have no dynamics.
• The bass doesn’t have punch. You boost it with some low-end EQ, but that just sounds louder and muddier. Not punchier.
• You can hear all the instruments in your mix, and they all seem to have their own “place” in the stereo image, but the overall image sounds wrong. Your other CDs have width and image that you can’t seem to get from panning the individual tracks.
• You had reverb on the individual tracks, but it just sounds like a bunch of instruments in a bunch of different spaces. Your other CDs have a sort of cohesive space that brings all the parts together. Not like rooms within a room, but a “sheen” that works across the entire mix.

Don’t worry. It’s not that you’re doing anything wrong. There are just some things you still need to do to get that “sound”. You just need the right tools and an understanding of how to use them. You won’t become Bob Ludwig overnight (or probably ever) but you can make dramatic improvements in your master recordings with a little work.

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This is a trick that you can use with any quantizer that provides the ability to apply a range of strengths of quantization. In Reason, there is a dropdown bar next to the quantize button that gives you a list of percentages ranging from 5% to 100%. What we are going to do is create a length of triplet notes on a hi-hat or any instrument, actually. In this case, let’s use 4 bars. For each bar, you should have 12 notes (the grid on the sequencer is labeled 1/8 T). And it helps to accent (apply a higher velocity to) to the first of every three notes. This creates a more natural, less robotic sound. For the first bar, leave the notes as triplets. On the first half of the second bar, apply a 1/16th quantization at 5%. On the second half, quantize the notes at 10%. Then at each additional half-bar, you will apply 25, 50, 75, 90, and 100 percent quantization. At 100%, you will have groups of three notes that are aligned to the 16th note grid, which is 16 notes per bar. It sounds kind of like a shuffle. You can also just leave out the 100% quantized triplets, and just fill in more bars with 16 notes per bar, with the first of every four notes accented. Listen to the included demo sound to hear this effect.

You’ve all heard it… a low-pass filter on the master out.

Here’s how to do it in Reason: Create an ECF-42 filter unit, and wire your mixer through the filter into the audio interface. Right click (Ctrl + click for mac) on the device and choose Create Sequencer Track for name of device. When your track is playing, hit record and turn up the FREQ knob to make it sound like you’re going back into the club. Try experimenting with the modes on the filter. BP 12 means band-pass filter. It allows only frequencies a bit lower and a bit higher than the frequency specified by the cutoff point (FREQ knob). The “angle” at which frequency amplitudes drop off is 12dB per octave. The LP filter has two modes: 12 and 24 dB per octave. The 24 cuts off frequencies above the cutoff point much more dramatically then the 12. The 12 leaks more high frequencies than the 24.

If you think of a song as an arrangement of layers of audio, then it makes most sense to begin a track with one or two layers. Instead of jumping right into a beat, it helps to ease the listener into the song. The intro gives the listener an idea of what the rest of the song will be like. It sets the mood, and at raves or other dance parties, it gives dancers a chance to catch their breath and rest for a bit.

If you are writing a downtempo or ambient song, it’s good to start the track out with sound effects with lots of delay on them. Try finding some sounds from the special effect presets on your synth. You want to ‘hook’ the listener from the first few seconds of your track. I can’t tell you how many producers start their tracks with a basic drum loop that stretches on way too long before anything interesting happens. In my opinion, that’s a very boring way to start a track (ok, it can be helpful for DJ’s, but still…)

In Reason, put a delay device on any drum beat or instrument. Create an automation track for it. Then, make sure that the track’s record toggle button is on. Hit record, and go wild with changing the delay time (the box with numbers in it), feedback, delay unit (ms or steps), etc. If done right, you get a crazy effect because a non-fixed delay time has to compensate for the change by fluctuating the pitch, it’s really weird sounding. Try it!

Here’s a handy PDF chart that will show you the frequency ranges of different instruments, and how to EQ them to achieve certain sounds. Includes: Kick drum, snare, hi hats/cymbals, bass, vocals, piano, electric guitar, strings, and acoustic guitar
(from Computer Music Magazine)

Imagine each moment of your stereo track as a box with three dimensions: Panning (left and right), Amplitude (volume) & Frequency. This box represents any given moment in time. Keep in mind that each instrument has a unique sonic footprint, or is composed of certain sets of frequencies. These varying frequencies can range in how far they span across the spectrum. In order to achieve a full, loud sound like the pros, you must fill this box in each dimension. The most common thing that prevents amateurs from getting a full sound is not filling this box properly. They combine sounds that overlap each other too far, which forces them to lower the volume of the song to prevent clipping (remember, the height of this box, or the volume/amplitude, cannot exceed a certain limit. Once it does, you will get distorted sounds, this is known as clipping). In order to maximize volume, it’s necessary to designate space inside this box to each element so that they all fit nicely together.

For a more detailed tutorial on this subject, check out this tutorial by tweakheadz

After hours of listening to the same track over and over, your ears will probably get exhausted. You will begin to lose your ability to mix the track properly. Take a break from your mix if you can. Sleep on it, and come back to it a day or two later. I guarantee you will hear things or notice things you wouldn’t have before. There’s nothing that can help you mix a track quite like distancing yourself from it for a while.

Have you ever wanted to make a knob move around on its own? With the Reason 3.0 addition of the combinator device, this is possible. And it’s pretty entertaining to see the MClass EQ device’s EQ curve dance around (as seen in the animation) when its knobs are moved automatically.

First, create a combinator device in the rack.
Then right-click inside the combinator device and create a Subtractor synth.

Press tab to flip the Subtractor around, and draw a cable from LFO 1 (under Modulation Input) to Rotary 1 on the Combinator device. Optionally, you can change the amount on the CV amount knob next to the Rotary 1 input, and this will affect the amount of modulation. Flip the rack around again, and click “Show programmer” on the Combi device. Select Subtractor 1 on the leftmost list, and on the right, select “Filter Freq” from the first list box. The boxes to the right control the minimum and maxium values that will be sent to the filter frequency slider. Notice the Filter frequency knob is moving on its own! Now, the section on the Subtractor labeled “LFO 1″ is where you can change the speed (rate) of the modulation. Notice the Amount knob doesn’t affect how the filter frequency slider is now moving. The LFO can still work independently and modulate the available parameters (like Osc pitch, FM amount, Phase, etc.) Try experimenting with the value of the Rotary 1 knob, the Min and Max boxes in the programmer, and the CV amount knob to get the perfect settings. Go crazy with the modulation routing, and animate different knobs, it’s great fun. Notice you can use ANY Modulation Output source and control the knobs like this. Anothing thing to try is the Spider CV merger/splitter. With this tool, you can send multiple CV signals all from one LFO. To make the animation seen above, I used the CV splitter to make two copies of the LFO signal, one of them inverted. That’s how I got one EQ gain paramater to move up while the other moved down.

A really mind-bending technique in reason is to automate the various knobs on a master auxiliary effect. To do this, set up an auxiliary effect (such as the ping pong delay mentioned on this site) by right-clicking on the main mixer and then creating an RV7000 reverb unit. This will automatically create the auxiliary send for you. Now, you turn up the auxiliary send knobs for each track you want to have an effect on. Then, create a new sequencer track and assign it to that RV7000 unit. Next, arm the track for recording and then press record. Now fiddle with the any of the knobs on the RV7000 and it will all be recorded. For an example of this technique in action, check out this excerpt from a track by Takyon called Hypergate:

Download Hypergate reason file (1.7 Mb)

This involves drawing many sequential ON and OFFs in your volume automation. For an example of this technique, listen to this example from “outer shpongolia” by shpongle. In Reason, there is really simple way to achieve this effect with the matrix pattern sequencer.

Here’s an example that you can use in Reason 2.5 or above: Download the example .rns file

Route the gate cv out of a matrix into the amp level in of any device. As you can see on the front side of the matrix, there is a note sequencer and a gate sequencer directly underneath it. Notice in the screenshot I have drawn in a pattern into the gate sequencer. This will automatically control the volume of the Subtractor synth. To create stuttery sounds, play a sequence through the synth while the matrix is on. Start drawing gating notes and see how it sounds. To create tie notes (the wide red columns that take up a whole grid box to themselves), hold down shift while drawing notes. Notice that you can also change the time resolution on the very right knob. This will make your pattern be interpreted at different speeds. For super-stutter, make it really fast =D Now you can program many different patterns by playing with the buttons on the left of the device. You can then automate the changes of the patterns for creating a unique, easy to manage microedited sound! For all other programs, you can automate the volume with a tempo grid turned on. Just draw volume automation into on and off patterns at different grid divisions.

This comes in really handy when you’re editing a drum track in Reason, in fact, it’s pretty essential. When you hold down shift while using the pencil tool in the sequence view, only the notes that you have selected will be affected. So for example, if you want to make a snare roll with a bunch of sequential 16th notes, but you have a kick drum hitting at the same time as some of the snare hits, you use the arrow tool to select just the snare rolls, and then use this method to alter just the snare notes, just draw a diagonal line that moves from low to high, and voila, a snare roll! (Snare rolls are even better, though, with a slight accent in velocity every 3rd or 4th note.)

Sometimes it’s good to have your music do things a human normally could not. The technique I call microediting has gained much popularity in the past decade, and it is featured in music such as Aphex Twin, Squarepusher, Mum, etc. To get that stuttery, robotic, glitchy sound, you have to zoom in really close on your notes. 1/64 is a good resolution to view at.

Here’s a small clip of a good example of microediting:

That second microediting part can be seen in the screenshot. Try setting your snap resolution to 1/64 and drawing sequential 64th on/off bars in your master or lead part volume automation. You can create really interesting, rhythmic stutter patterns this way.

One of the most popular reverse hits is a reverse cymbal. You hear it all the time in pop or techno songs. The reverse cymbal crash leads to the next bar. It works well as a buildup and transition. To make your own reverse hit, try taking a nice loud percussive sound that hits and then gets quieter from there on out. Then in any sound editor such as Audacity, reverse the sound. Save it and import it into your sequencer with a sampler. Then experiment with the amount of time you have to play the sound so that it reaches its peak right at the point where the next bar begins. Check out this reversed sound of a metal barrel being hit (I use this one all the time as a precursor to a huge transition).Example 2: a cool free reverse hit on freesound

This is a widely used technique for giving an instrument more life, and for creating buildups. In most software sequencers, you can automate knob settings over time. One of these is the cutoff frequency of a filter. Try using a low-pass filter or high-pass filter on a sound and automate the cutoff frequency.

Check out this example from “echonomix” by infected mushroom. They are using a high-pass filter and automating the cutoff frequency here.

The best buildups I can think of are by Infected Mushroom. They use many of the techniques described in Edits & Efffects to create tension before a beat busts in and you just HAVE to dance. Buildups will combine many of the following tricks all in one go for full effect.

Listen to this example from infected mushroom’s sailing in the sea of mushroom. Notice all the different components used in this segment to create tension. They used a very long reverse hit, they introduced new, different layers each bar or so. They also automated the cutoff filter of the lead gated synth. Then when the buildup reached its peak, everything was silenced as a drumloop kicked in and another shorter reverse cymbal played. Then it reintroduces the full ensemble of layers for that full intensity effect.

In most sequencers, there is a quantize function. This will take all of the note data on a track and aligns all of the notes to the nearest division. So if you play sloppily or you just have bad latency on your midi keyboard, quantize is your friend. Select the division most appropriate to what you’ve just played. 1/8th and 1/16th divisions are most common. Another cool thing about quantization is that it will sometimes snap a note to a division that you didn’t intend. But it sounds good anyway, minus a few odd parts. Then this quantized melody may inspire you to come up with a new melody altogether. Of course, not all music sounds good with quantization. If you’re going for an organic, human sound, then you probably won’t want to quantize 100%. There are options in most sequencers to quantize to a certain percentage. This is useful if you want to correct your timing, but not to the point of computerized perfection.

So, you don’t want your melodies to be boring? My first suggestion would be to come up with different variations on a melody that you can use as a loop. Then vary which loop you use in the sequencer. For example, if you wrote two melodies (we’ll call them A and B) and then made a variation on B (call that one C), you could sequence A B A C. This is a very common pattern. Or perhaps try A A B B, or A B A B, etc. My next suggestion is automation of several different knobs on your synth. For example, you can automate the cutoff frequency, vibrato, volume, modulation depth, etc. One trick I commonly use on my synths is to route an LFO to the filter cutoff frequency. Then I route the modulation wheel on my keyboard to the LFO’s depth (the LFO speed should be synced with the tempo at 1/8th or 1/16th). You could also route the modulation wheel to the LFO’s speed for an interesting combo-modulation. The modulation wheel on any midi keyboard is handy for live performances. But when it comes to sequencing, there are no limits to how many automation tracks you can use. I suggest automating many parameters at once to keep things interesting. For an example, listen to the solo synth on my track, “your best shot”. In this track, I am automating the cutoff frequency for those intense moments (on a low-pass filter), amount of vibrato (depth of LFO routed to cutoff frequency), and amount of delay (via the aux send knob). All of these combine to make for an expressive, almost human-like quality to the synth.

In this screenshot of my song, (made with Ableton Live), you can see the automation of the vibrato amount overlayed on top of the notes.
Listen to the melody shown in this image.

The vertical lines in a sequencer are what slice up musical notes in time. Depending on what resolution you are viewing your sequence at, there can be many variations between how closely the vertical lines are spaced. But before we can look at the divisions of time, we must talk about time signature.

The time signature tells you how many beats are in a bar and what note or rest is equivalent to one beat. Most time signatures are 4/4, but things can get interesting when you change the time signature. Whenever the second (or lower) number of a time signature is 4, this means that one beat is equal to a quarter note. When the second number is 8, one beat is equal to an eighth note.
Get a more detailed explanation.

For a 4/4 time signature:
1 bar = 4 beats (4 quarter notes) = 16 sixteenth notes = 32 thirty-second notes
As you can see, there are many different resolutions that you can split 1 bar into. Typically, resolutions of 1/32 and 1/64 are in the realm of microediting because any change of notes at these divisions will sound very quick.

Say this out loud to get an idea of what 16th notes are like:
“1 e and a 2 e and a 3 e and a 4 e and a”
This whole phrase is equal to one bar. Each utterance equals a 16th note. Count ‘em up, you’ll see that there are 16 separate divisions.

Vary your note lengths: If you want your melodies to be expressive and interesting to listen to, you must vary your note lengths. After drawing a sequence of notes, try altering the length of the notes and shifting them from the left or to the right in the sequencer.

Vary your note velocity: Same as if you’re programming a drum track. Unless you specifically want a track to sound mechanical and computerized, you should randomize your note velocities a bit. All the while, loop your sequence and listen to it as you go. You will get sick of it eventually, and that’s when you know that it’s time for a break.

Sometimes it’s just easier to draw melodies in a sequencer than it is to actually play the melody on a keyboard. Especially if your music is fast. Essentially what you will be doing is drawing a rhythmic pattern composed of note segments that can be organized within a scale. Most software sequencers have 2 dimensions: The horizontal, or time dimension, and the vertical, or pitch dimension. As the playhead progresses along from left to right, the notes that are drawn on the horizontal lines are played. The rest of this section will be divided into two parts, one for each dimension.

The first thing I do is create a new Subtractor synth. By default, the selected waveform is a sawtooth wave (the sawtooth wave just happens to be a nice waveform for a cool bass sound. Also by default, in the lower right corner of the synth, the velocity is mapped to F.env (filter envelope). By this default, when you hit a key harder, it will sound more bright. You will briefly hear these higher frequencies before the filter cuts them off at the speed set by the attack and decay on the filter envelope.

Next, I change the polyphony to 1 instead of the default 8. That way, if I press more than one note at a time, only one will be allowed to play. For the most part, bass instruments do not play chords or simultaneous notes. Next, I lower the octave on Osc 1 to 2. You will want to lower your octave on any bass patch to get a nice deep sound. Then, I change the mode of the oscillator to X (there are 3 options directly to the left of the waveform selector: X, -, O) What this does is creates a duplicate copy of the waveform and stacks it on top of itself. Then you can control the “alignment” of the two stacked waveforms with the phase knob. With the X mode, the waveforms are multiplied, and create a nice fat sound. If one waveform is subtracted from the other (- mode) then it will sound weaker. If O is selected, then a duplicate waveform is not created at all. So for a thick juicy bass patch, I leave the mode set to X and shift the phase knob to my taste.

Next, I decide how I want the filter envelope to sound. If I increase the amt (amount) knob, the filter envelope will be more noticeable. Then I can create a sort of fade-in effect on the filter by increasing the attack time. This is nice for slow notes that build in intensity as time passes. If you only want this fade-in effect when the keys are pressed lightly, and not when you hit them with force, you can also change the amp (volume) envelope. Increase the attack time and then turn the A. Atk knob in the Velocity area to the left. Now when you press lightly on the keys, the sound will fade in, and when you press hard, it will snap right into action.

Finally, I add EQ and Compression (not the dinky old kinds, I’m talkin’ about the MClass units). Emphasize those low frequencies with the EQ to really make your subwoofer rumble. Then use the aforementioned compressor settings to make sure the bass cuts through all the other parts.

download the example RNS file

Do you have a problem making your basslines thump in different audio systems? Do they sound nice on one set of speakers, but weak on the next? Are you having a hard time getting the power you want from your basslines without making the mix too muddy? Here are a few tips on how to make your basslines bang. (from Future Producers)

The Frequency Range
If you want your bass to bang in a system with nice subwoofers AND in crappy home shelf systems, it is pointless to use a bass patch whose energy lies only below 40 Hz, because most home systems will not play sounds that low in frequency. You need to make sure bass has a lot going on in the 70-90 Hz frequency range. So just how do you do this? How do you get a sound that is both felt and heard on a number of different speaker systems?

Layering Other Waveforms
The sine and triangle wave produces that low thumping bass tone we electronic composers love (e.g. sub bass, 808 boom, DnB drone). These waveforms have few or no harmonics, so they are felt more than they are heard. If using a synth (or even a sampler), try layering these waveforms with a waveform rich in harmonics, such as a square or saw wave. After layering, use the synth’s or sampler’s low pass filter cutoff to trim away some of the higher harmonics from this new bass patch.
Read the rest of this entry

For those times when your loops in Dr. Rex are twice as slow or twice as fast as your song, there is an easy way to slow down the loop or speed it up. In the sequencer view, select a group of notes and right click on one of them. Then select “Change events”. You will see an input that is labeled “Scale tempo”. If you want to double the speed, type 200% and then press ok. Otherwise, type 50%. You can experiment with this for interesting loop speed variations

When you use the Dr. Rex loop player, you can easily change the pattern to break the monotony of looping something over and over. Here’s how: on the Rex player, there is a button labeled “to track”. Press that, and notice that in the sequencer view, the space between your left and right loop points will fill up with colored boxes. Right click (or ctrl+click for mac users) on one of the colored boxes (this is a group of notes), and then choose “Change Events” from the pop-up menu. This gives you many options for changing a group of notes. You want to use the “Alter Notes” functions. This will randomize the notes according to whatever percentage you choose. Now listen to the loop, it will be remixed. Granted, it is somewhat randomized, so you may have to switch over to the note view and change it to your liking. But this is a really great way to come up with drum tracks in a matter of minutes. Try it on a whole bunch of note groups all at once and listen to all of them. Chances are, there will be some remix of the notes that’s a real keeper.

“The fatness of the snare tends to reside between 120 and 400Hz. A boxy sound is indicative of comparatively high energy in the 800Hz-1.2kHz range, whereas the resonances of the drum’s ringing reside above this, between about 2-4kHz. The crispness of the drum’s attack tends to reside more in the 4-8kHz region.”
(from Sound on Sound)

A well-known trick to beefen up your snares is to combine two snares with complementary characteristics. For example, combine one snare that has a nice snap to it with another snare that has a thump to it. Optionally, you can EQ the snares to accentuate these characteristics. EQ the lower pitched snare with a lower EQ and a higher EQ to the one with more snap to it. This will result in a slap-you-in-the-face snare sound.

Try adding a little bit of delay to your snare. A delay of 3 steps will give you an interesting syncopated effect. You can alternately use a delay with a setting of just 2 steps for the dub/reggae feel.

If you don’t want your hi-hat track to sound mechanical and lifeless, then you need to vary your patterns. This can be done with a few different tricks. First, alter the note velocities to create accents as seen in the left image. A good starting point is accenting the first of every three or four hits. Make the velocity on these hits higher than all the other hits. Second, set the hi-hat’s volume envelope to the following settings (and as seen in the right image). Attack = 0, Decay = just above 0, Sustain = 0, Release = 0. Now you can automate the decay time to make the hi-hat hit shorter or longer. This emulates the sound of a hi-hat being closed tighter or looser.

“The punch component of most bass drums lies between about 80 and 100Hz. Below this area, you’ll mostly feel, rather than hear, any boost, and it’s easy to overdo. Warmer kick sounds major on the 200-300Hz region. When the kick needs to cut through on smaller speakers, then you might also consider a boost in the region of 2.5-6kHz, which will tend to emphasize the click of the beater.”
(from Sound on Sound)

If you want a dull sounding kick for mellow tracks, you can apply a lowpass filter to it to create a low thump with no high-end whatsoever. For trance and breakbeats, it’s good to add some high frequencies to the kick so it has a little bit of click to it. This will give the kick much more presence and make your track more danceable. You should apply two separate EQ curves to the kick to achieve this. One EQ curve should emphasize the low frequencies (which usually are just below the bass frequency range. you don’t want the kick and bass to reside in the same frequency because that will result in a muddy sound) The other EQ curve should be just above the bass frequency range. The combination of these two EQ curves will give you a “saddle” for the bass to sit in.

If you’re like me, you like songs that can tell a story, even without words. In order to create a song that can tell a story, it must be able to communicate different emotions throughout the length of the song. This means that your song may contain several different themes all together. So, your song will need to be able to smoothly transition between the different elements. I find that the easiest way to accomplish this is to gradually (or suddenly) drop out most of the layers in your track, leaving only one or two different sounds or instruments. Then introduce a new element that sounds good with the last remaining layers, and use a buildup to introduce a new section. Now you can drop out those last remaining layers, and now you have a completely different vibe!

Another interesting trick is to find a nice long percussive sound and reverse it. Then, when you have the reversed clip in your sequencer, you align the climax of the reversed sound with the beginning of the next bar. This adds suspense, which is cool in my book.