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.
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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.

Start off by creating an RV7000. Now personally I think you should bus it to your main mixer on one of the aux tracks. To do this, right click on the mixer and create the device, it should automatically attach the device as an auxiliary effect. This will allow you to send the audio signal to the effect with the aux send knobs on each mixer track. This will give you control over the dry/wet sound, and you can apply the effect to any track on the mixer. I would not recommend attaching a space/delay type of effect directly on your device; this reduces the versatility of the effect. Okay, now that that is taken care of, expand your RV7000 view by hitting the carrot on the left [it’s the triangle pointing at the remote programmer, right under the patch loading box]. With this open you can now adjust the parameters of the device. The algorithm is set to hall, but we want multi tap, so adjust the top knob on the left side till the algorithm reads “multi tap”. Alright, now notice the knob second from the top on the left side. It should be set to tempo sync. By default, it it should be set to OFF, but we want it ON. With this on, the delay will sync to the tempo that you have your track set to. Set the diffusion to about 15 (second knob from the bottom on the left side) and the LF damp about 385Hz (bottom knob on the left side).
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