Sub Bass Triangulation...

Correct EQ-ing of the low frequencies of a song requires decent speakers and room acoustics. Even though we covered some mixing and mastering tricks before, sometimes we might find ourselves in situation in which we have to be away from home for longer periods of time, without access to our main gear.

If you have portable speakers and headphones that you can trust but you are unable to use them for detection of very low frequencies, there is still thing that you can do. Just make a combinator patch and insert it in the end of the audio signal chain. 

The signal inserted in the combinator goes into the first channel of a Line Mixer. An NN-19 device with a Matrix Pattern Sequencer attached to it goes into the second channel of the mixer. 

The output of the mixer goes into a Stereo Imager and the output of the imager is routed to the output of the combinator. 

The next thing you need to do is to find some commercial song which sounds the most similar to your track. Crop some short part of it`s drop and insert it as a sample in the NN-19. 

Set the separators of your track to loop the drop of your song. 

Use the first combinator assignable button to setup channel 1 of the mixer to solo when pushed on and channel 2 to solo when pushed off. This way you can toggle between the loop of your song and the loop of the sampled track easily.

Use the two combinator assignable knobs to manipulate the volume of those two mixer channels. 

The stereo imager in the end of the chain should have crossover at about 150-200 Hz (cutting at the low end you have trouble to hear). Use the second combinator assignable button to solo / unsolo the high frequencies only.

The idea behind the approach is to cut the low frequencies and then to scale the volume of the two tracks using the big VU meter as a helper.

Once the volume has been scaled, you can unsolo the low frequencies and see whether you have put too much sub compared to a professionally mastered track.

Note that this is only a rough approach.

Mixing and mastering tips

This post will contain some healthy tips when it comes to mixing and mastering with Reason. It will be useful to people who record music with non professional equipment at home, trying to achieve result closer to the professional.

The minimum requirement when it comes to equipment is a set of decent speakers with decent bass response. For the subtleties shaping, the cheapest thing I can recommend is using good pair of headphones (however, if conflict arises, the speakers should have the final say). You need to spend time getting to know your sound devices by listening to commercial tracks similar to the ones you are planning to create. 

If you are like me and neither have good room acoustics, nor flat studio monitors, you can at least get some decent flat response mic and equalize your room by playing pink noise and then using spectrum analyzers on two channels  - one listening to your PC internally and one to the signal coming from your mic. Try to make the second one`s behavior the same as the first one with equalizers as much as possible. 

If you don`t have acoustically treated room that gives you correct representation of the sub bass, one thing you can do is to take a little bit of shortcut and touch the sub woofer cone, feeling the vibration of the bass that way.

I divide a track into three elements - percussive mix (drums, snares, hi hats, etc...), melodic mix (leads, basses) and sub bass. In the beginning I mix the first two and when I make them sound somewhat decent together I add weight by the sub bass tones. The sub should not be either too quiet, nor too loud drowning the mix. It is a good idea to clear all the unnecessary frequencies from each sound, even those which you might not hear, but can see on a spectrum analyzer. Some elements require either boosting or reducing specific frequencies. Generally, when boosting frequencies, you create a boost with wider Q, in order for the result to sound more natural, while when cutting, you use thinner Q. If there is some tedious frequency which you want to remove, but having trouble finding, you can make a big boost with thin Q and begin searching where it is trough the spectrum and remove it completely when you find it.

In the global EQ on the mastering stage, you may have slight boosts in the low end and in the high end, but no more then 3db. When you use the stereo imager, you generally need to make lower frequencies more mono, while making the higher ones wider. Compression and limiting amount depend on the genre and taste of the producer - some prefer more dynamics in their tracks, while others prefer LOUDNESS. Generally, you need to boost no more then 6 db above 0 dB when you master the track. You also need to work gently and precisely and not to quicken things up.

You can also check how your track sound in mono if you have trouble deciding how loud some expanded parts of the mix should be, or just to have an idea how it would sound in such devices. You can also solo the frequencies above 200 Hz to have an idea how the track sound without the subs.

One way to clear up room in the melodic content of the song is to cut all the unnecessary frequencies which do not belong to the key scale of the song. Don`t do this on the percussive mix though, because those sounds are not supposed to be on the same key of the song necessarily.

Keys corresponding to frequencies chart.(Hz)

For example, if you use the C major scale which has all white keys, you don`t need the black keys sounding as prominent. Of course, do the cuts wisely - the mix should not sound unnatural.

If you have spare time, you can make yourself combinator patch containing EQ-s, set by a modifying rotary to have thin q cuts on all the minor keys corresponding frequencies. By moving the rotary, you can set the cuts` offset (the relations will remain the same) according to the major scale you are using and cut its unnecessary frequencies. You can create another one for the minor frequency relations as well. 

That`s basically it. Listen to your track relative to other similar sounding commercial ones and try to make it sound as close as possible.

Audio comparator patch

#8 MClass Mastering Suite Devices

The MClass Mastering Suite consists of 4 devices which you can use in the final mastering process of the mix to make the song sound better and louder, as well as to shape different sounds manually beforehand. 

Since there often is a problem when you play different instruments which manifests in the fact that they sound good when soloed out, while sounding bad when played together, you have to make certain edits on each individual one in order to make the mix sound right. That`s where the MClass Equalizer comes in handy...

It has five sections, each having three knobs, except for the first one. They are activated by clicking on the small squares above them.

The LO CUT section cuts the frequencies below 30 Hz, which are usually problematic and not commonly used in most of the music genres. 

The idea behind all the knob columns is basically the same - the one on the top selects the frequency, the one in the middle rather cuts or boosts that frequency and the knob on the bottom determines how wide that cut/boost will be. 

PARAM 1 and PARAM 2 are identical - they cover the range between 39 Hz and 20 kHz. LO SHELF and HI SHELF cover the ranges between 30 Hz - 600 Hz and between 3 kHz - 20 kHz respectively. A small picture on the bottom gives visual representation of the shape of the curve of each section.

It is irrelevant whether a sound sounds good or not after equalization when played by itself - the important thing is to sound good in the mix. That`s why it is important to make sure that you cut all the undesired frequencies (for ex. higher ones from bass frq, lower ones from treble ones, etc...), even those you can`t hear, but can see via spectrum analyzer. If a sound gets soloed on a certain moment of the track, you can use automation in order to temporary switch off the Equalizer, since no conflicting sounds are present at that time. The more time you spend removing undesired frequencies, the more free space you get for loudness boost in the final stage of the mix.

The Stereo Imager is the next device we are going to look at.

It is a device that increases or decreases the wideness of the sound in the stereo field. The sound is divided into low part and high part, the border of which is determined by the X-OVER FREQ knob in the middle, covering the area between 100 and 6 kHz with mid point of 787 kHz. If the knob for the low on the left and the knob on the right for high frequencies are at 12 o`clock, they are not effecting the sound. When twisting them to the left, you make the sound more mono, while by twisting them to the right you make ti wider in the stereo field (there are indicators giving you visual picture of what`s happening). Usually, bass frequencies tend to be more mono, while high frequencies are more expanded. It is a common practice to make certain frequencies below 200 Hz fully mono. 

With the buttons on the right you can single out either the low end or the high end out of the sound. This comes in handy when you wish to quickly cut undesired frequencies of a given sound.

You can see that on the back side of the device there is one extra output, which solos the low or high end. You can use that, combined with the solo option on the front, in order to divide the sound into two parts, linking them to two separate mixer channels and then processing the sounds differently afterwards. 

Next comes the Compressor. As the name suggests, it is a device compressing the sound, making it sound fatter. 

If you use the left INPUT GAIN knob, rather then the right OUTPUT GAIN one, more of the sound gets processed by the area in the middle. The loudness of the track required in order for the compression to start compressing is determined by the THRESHOLD knob, covering the area from -36 to 0 dB. 

Hard/soft knee visualization... 

Once the loudness of the track reaches the threshold level, the remaining volume area above it gets compressed by the ratio determined by the Ratio knob, starting from 1:1 (no compression), 2:1 cutting it by half, ... , ... and infinite in the end cutting out everything above that point, acting like a limiter. The indicator on the right of the knob shows visually how much of the sound vol is reduced. 

The compressor also has attack and release option. The attack determines how fast the compression starts to kick in after the threshold level has been reached. The more you turn it to the left, the faster the compression begins to affect the sound, while the opposite happens when you turn it more to the right. The release on the other hand, determines how long will it take for the compressor to "release" it`s compression when it falls down below the threshold (it affects the sound below the threshold with the same ratio). Here is visualization of the behavior of those controls:

In general, if you have more compression, you will need quicker attack and longer release in order to avoid pumping and breathing unnatural effects. 

In order to adjust the attack of your drums properly, make extreme compression my lowering the threshold at - 36 dB with infinite ratio, make sure that the soft knee is off and the attack and release times are at minimum level. Then, move the attack knob up and down until you find the clicky sweet spot. After you adjust the attack, do whatever compression amount you like.

If you have no idea what the release amount should be, try something closer to that formula:

(60 / tempo) / 4

Example tempo 170 bpm. 

60 / 170 = ~ 0.350 ms   (1/4 note time)

350 /4 =  87.5 => use 88 ms (the closest to it value that the compressor can have)

The basic explanation of the adapt release function is that it will make the release affect shorter peaks and automatically adjust when longer peaks are present in the future. 

This compressor has also got sidechain input trough which it receives information on the loudness of another device, and makes the compression amount based on it. For example, such compression with quick attack is useful for certain types of dance music, where the volume of the bass is quickly reduced when the kick is being hit and then regaining its loudness immediately after the ending of the hit. The Gain Reduction CV Out, as the name implies, creates CV signal controlled by the gain reduction (visually seen using the indicator on the front). 

You can solo the signal fed into the sidechain input with the solo option in the sidechain area on the frond. The ACTIVE indicator shows whether there is device connected to the sidechain input or not.

The MClass Maximizer is a device that raises the apparent RMS loudness of the mix by cutting the peaks (freeing space that way) and then boosting the processed signal. It also prevents clipping. Although it is a device which is always placed in the end of the mastering chain, it is common practice to start adjusting it first.

The device has, like the compressor, input and output gain knobs, serving the same purpose. The LIMITER button prevents the audio signal from going beyond a certain loudness level, by cutting it, similar to compressor with infinite ratio.  It also has gain reduction indicator. 

The 4ms LOOK AHEAD button will create a 4ms delay (more on delays in later tutorials) which basically detects peaks before they are being played back, so that the limiter is prepared for them. (the OUTPUT VALUE knob works when it is switched on).

The attack and release options work in a similar way to the compressor.

0 dB is the maximum level of the loudness measuring scale in the digital world (the higher negative number you have, the quieter it sounds). Cranking up the volume above that level results into distortion.

Gradual tendency in music of abusing the limiter more and more trough the years...

To make the sound apparently louder, the limiter cuts the peak points by the percussions and then cranks up the volume of the track filling the freed space. You basically sacrifice dynamics for volume. Pay special attention to the loudest parts of the song when using the limiter.

When you switch the limiter on, switch the 4ms LOOK AHEAD on, set the attack to FAST and the OUTPUT GAIN to 0 dB, you`ve created a "brick wall limiter", which will not let any sound pass trough the 0 dB level. 

The SOFT CLIP option also acts like a brick wall limiter, however instead of making hard clips or cutting rudely parts of the transients, it adds much more pleasant soft clips and rounds things up. The amount of the soft clip is determined by the knob on the right.

The difference between the two types of meters on the right is that the peak has faster response, while vu gives an average reading.

That is basically all you need to know when it comes to the mechanics of the mastering suite.

The PEQ-2 equalizer and the COMP-01 compressor are two oldskool devices, derived from older versions of Reason and kept for compatibility purposes. They can be used as CPU saving devices, when you don`t need the precision of the MClass tools. However, unlike the MClass one, the PEQ-2 eq has two frequency CV inputs, .

Reason has a big meter which you can use for more detailed analysis of the sound dynamics. Just select Big Meter from the Hardware Interface and then click on the little square below the input or the output channel you want it to monitor (usually it is the first output channel). You can also do it by rotating the CHANNEL rotary of the meter.

There are 3 different light indicators which each channel can use - green means that the channel is in use, yellow - that it is available, but not in use and red - that it is connected, but not available. No light indication means the channel is not available.

You can select from 3 different indicators or combination of them by clicking on MODE.

VU (volume unit) - it uses the blue measurement system. It is a slow meter, showing you the average sound volume, emulating the human ear sound perception. When it reaches 0 dB, it colors the volume above it with red. The offset of that point can be changed via the VU OFFSET knob.

PPM (Peak Programme Meter) - It and the peak one below use the orange measurement system. It displays the peak signal volume and has very slow release.

PEAK - It shows the peak volume of the signal. It has very fast attack and release. The measuring indicator becomes yellow above -12dB (same as the PPM). This meter holds the detected peak volume indication for either 5 seconds or infinitely, determined by the PEAK HOLD button. The indication can be reset at any time with the RESET button on the right.
 

Here is an article with some tips on mastering.