MIXING & EFFECTS
DEMO ONLY: Maximus comes as a demo version in FL Studio and needs to be purchased separately so you can save projects containing Maximus effects.
This page has two sections, the first section explains signal routing possibilities in Maximus while the second section delivers a series of tutorials spanning the basics through to some of the more advanced processing techniques. If you want to learn how to use Maximus, this is the place to start.
The input signal may be Lo cut (LO.CUT, Band Frequency Controls) to remove subsonic energy, improving headroom for the remaining 'audible' frequencies. The signal may then be (optionally) up-sampled (General Options , lower Analysis Display), to improve the quality of the next stage - band-split filtering, into LOW, MID and HIGH channels. The LOW/MID and MID/HIGH boundaries are user-variable, according to the LOW and HIGH controls (LOW FREQ, HIGH FREQ, Band Frequency Controls) as are the band-split filter types (IIR or Linear-phase, General Options ). Each band can then be processed according to -
Once processed, the signal is then recombined and (optionally) down-sampled (General Options), as it passes to the final MASTER compression stage. The same processes used on the frequency sub-bands (as described above) may be applied. Finally, the signal is sent to the plugin outputs and onto the operator's ears where it is met with large nods of approval.
The signal flow chart does little to explain how to use Maximus so we have provided some tutorials to help you understand the diverse capabilities afforded by Maximus. This section covers -
NOTE: The key parameters to the 'sound' of most compression settings are the PRE and POST gain, compression envelope shape and the RELEASE time. You will need to fine-tune these parameters to suit the timing and dynamics of the sounds you are Maximizing. Please also note that the tutorials are a guide, there are no absolute right and wrong uses for Maximus, if something sounds great it is great. Now, if only we could all agree on what 'great' sounds like.
Compression is a form of automated volume control, that reduces the dynamic range of sounds. When the input signal exceeds a predetermined threshold the gain is reduced. This does not mean the sound gets quieter, but the rate at which the waveform is increasing in loudness is slowed or stopped altogether. In other words, the volume is held within a narrower range than it originally occupied. The art of setting a compressor is in fine-tuning the magnitude, speed and timing of the automated gain changes so that the compression process does not introduce artifacts.
So, how can reducing the maximum amplitude peaks of an audio waveform make the sound seem louder? To understand we need to consider the way our hearing interprets the start (attack) and body (sustain) portion of sounds. It transpires that the attack (the first ~50 ms) is used mainly to form impressions of timbre, clarity and crispness, while the sustain (the sound that follows) contributes most to the perception of loudness, pitch and 'punch'. The sustain has most weight because our perception of loudness comes from an integration (adding/averaging) of the preceding 600-1000 ms to any given moment. Attack transients, simply because they are of very short duration, don't have as much weight as the main body of the sound. It is by lowering (compressing) the maximum amplitude of the transient peaks that we free up headroom to raise the gain of the sustained portions of the signal (post compression 'make-up' gain), as shown below.
1. The uncompressed input is 2. compressd so the transient 'attack' peaks are reduced, then 3. the level of the sound is increased (often by the same amount as the compression reduced the peaks). It is this step that increases the loudness and punchiness of a compressed signal. However, as we alluded to earlier, compression represents a trade-off between dynamics and loudness, welcome to the loudness wars!
Maximus has 4 independent compressors and compression envelopes, one for each of the LOW, MID, HIGH frequency bands and a MASTER applied to the combined output of the three sub-bands. Why 3 sub-bands? When compression is applied to a single band across the whole frequency spectrum, a loud sound (bass-heavy kick drum, for example) will trigger the compressor, decreasing all frequencies, including the high frequencies. This may be heard as a fluctuation of the track's brightness in time with the kick, and is known as 'pumping' - desirable in certain styles of music, but not generally considered a good thing in a final mix. Isolating frequency bands, such as the bass and treble, and compressing them independently, avoids pumping effects and so allows for a more 'transparent' compression, without artifacts.
So how do you control the compression parameters in Maximus? If you have some experience with traditional compressors you may have noticed that Maximus does not have threshold level or ratio controls. Instead, the threshold and compression ratio are fully customizable using a user-definable envelope. The graph represents the input (horizontal) to output (vertical) relationship and is described in more detail below.
In the left example 'No compression', input levels are passed unchanged to the output, regardless of the level. Note that the middle and right graphs also have input ranges, where no compression is applied, prior to the points marked 'Threshold'. The middle graph, 'Heavy compression' shows modification of the output starting after -3 dB, the point where the curve deviates from the 1:1 (45 degree) line. The deviation point represents the compression threshold, i.e. the level at which compression starts. It is clear that as the input level increases past the threshold point, progressively more compression is applied to the input signal. This compression curve would be typically used with percussion sounds, to cut the peaks, and allow the remaining parts of the signal to be boosted without clipping. The right graph, Hard limiting depicts a special case in which the signal is prevented from increasing once the 0 dB threshold has been passed. Limiting is described in more detail in the 'Limiting' section.
The possibilities are endless! Here we encourage you to think about some of the more creative uses for the Maximus compression envelope. The left panel 'Gating' will cut any input signal below -12 dB. The 'Boost & limit' curve will amplify low level signals, then limit them to 0 dB, while the 'Inversion' curve will invert the volume of a sound - loud is quiet and quiet is loud. Indeed, anything is possible when you are using Maximus 'creatively'. It's worthwhile to take some time to experiment with the MASTER compression envelope of the 'Default' Maximus patch, to see how sounds are affected by various compression curves.
In this example we will learn about the MASTER compressor settings, and how to use Maximus as a single-band compressor.
In the analysis above, the release time was varied from the minimum (left side of the trace) to a higher value (right side of the trace). In this instance the longer release results in a smoother curve which closely follows the input signal without being affected by the high frequencies, and so is probably a better setting. As usual, your ears should be the final guide for all settings, with the display analysis as a guide.
NOTES & TIPS: For the compressor to work the input signal must exercise those parts of the compression curve that modify the output level. This author has to admit on more than one occasion spending several minutes wondering why the compressor isn't working as expected, only to discover the input level wasn't exceeding the compression threshold (Doh!).
Limiting is simply heavy compression (generally used to describe compression ratios greater than 10:1). The purpose is usually to 'limit' the output volume at a certain maximum level, usually 0 dB, to avoid clipping in a final track. In this example, we will use an infinite compression above the limiting-threshold, i.e. no increase in output regardless of the input volume. Limiting prevents signal transients (fast peaks) from clipping the final mix. Although limiting may be applied to any channel in Maximus (LOW, MID, HIGH), we will concern ourselves with the complete mix and apply a Limit curve to the MASTER compression envelope and so limit the maximum output to 0 dB.
In the example above, the MASTER compression curve has been set to hard limit the output to 0 dB once the signal exceeds this level. Since the audio analysis takes a finite amount of time, it is possible that the fastest transient peaks will slip through the compressor before the system can react. To solve this problem, a fast attack and look-ahead are used. Look ahead applies a small delay to the plugin so that it can 'look-ahead' into the signal to detect the fast transients and apply the compression 'just in time'. For complete mix limiting it is less critical to keep the LOW, MID and HIGH bands below 0 dB, as the MASTER limiting will prevent clipping.
NOTES: Although it may be hard to determine from the peak-display, it is possible that the example used here has too much limiting for a final mix, since most of the input signal exceeds the limit threshold. This could be a problem, since much of the dynamic (volume) information of the track is being 'squashed'. Further, extreme limiting will introduce 'saturation' like effects that you may not want in a final mix. The solution is to turn down the band PRE GAIN inputs, so that the occasional peak is limited rather than most. Finally, spare a thought for others who may be working on your track in future. Are you passing it onto a recording engineer or a mastering house? If so, try to avoid heavy limiting or compressing of the final mix, since it leaves them little scope for applying the 'art' of mastering.
Parallel, or NY compression, is the term given to the compression technique in which a dry (uncompressed) signal is mixed with an over-compressed copy of the same signal. The purpose of such compression is to increase the average volume of the quieter parts of the signal, while preserving transients that would usually become squashed when heavily compression is applied. NY compression has a distinctive sound and can be achieved as follows:
If you have the settings correct, your percussion track should be transformed into a huge pumping wall of sound when you turn the LMH Mix control to the right.
Sidechain compression describes the process in which the sound driving the compressor envelope is (usually) unrelated to the sound being compressed. The term 'Sidechain' comes from analog compressors that accepted an audio-input separate to, and unmixed with, the signal chain. In other words, a 'side-chain signal'. The Sidechain was used as the control signal (e.g. kick drum) and the signal-chain was compressed in response to this sound. Side-chaining causes a pumping, ducking or gating effect, as used in the 'Parallel / NY Compression' tutorial above.
Strictly speaking, Maximus does not have a sidechain input, however there is a work-around as Maximus can selectively listen to a narrow slice of the frequency spectrum focused on the signal you want to control the compressor by. In this tutorial we will set Maximus into MID master mode, so that the MID band acts as a single wide-band compressor, and then single out the trigger frequencies in the mix using the LOW and HIGH cut controls on the MID band.
Noise Gating is the process of muting a sound, usually after the signal falls below a certain level. Noise gating is useful to mute low level hum or hiss, often found in recordings made at live gigs. In this example we will set the MASTER Band compressor curve to mute once the input signal falls below a certain level. As you work through this example, bear in mind you can also isolate the sound in one of the LOW, MID or HIGH bands and gate only the troublesome band.
Ducking is the process of momentarily lowering the volume of one signal in the presence of another and relies on the same principles as used in the side-chain compression tutorial above. Ducking is most commonly used in voice-over applications where a music track needs to be lowered so that the spoken part can be more clearly heard. Radio-announcers, DJ's, PA operators and movie-makers can all benefit from ducking.
De-essing selectively limits the high frequencies in a sound (usually vocals) from exceeding an acceptable maximum. Often, when recording vocalists, 's' and 't' sounds become overly bright and detract from the quality of the recording. Sibilance, the technical term, can be tamed by 'de-essing' the recording (so called as it affects the 's' sound most noticeably). The novice may simply turn down the high frequencies with an EQ, however this causes the vocal to lose its brightness and clarity. Here we will outline an approach that is based on selectively compressing the problem frequencies, and only when they occur.
So you've noticed that the input/output of Maximus is measured in dB (decibels) and wondered what it means? Consider that a signal passing through Maximus without change has a 1:1 input to output ratio (1*input:1*output). Since we are engineers, and therefore quite nerdy, we will decide that all levels are thus to be measured relative to 1 (no change). We also agree among ourselves that values below 1 are negative and values above it positive (think negative and positive change). At this point we also realise that fractions aren't exclusive enough (even our mothers will understand if we say the signal level is half as loud). What to do? Let's choose to work in a logarithmic scale and, since all values are relative, let's use a 'relative' logarithmic scale - the decibel! At this point we (being engineers), look around and realise that no-one knows what we are talking about and nod vigorously at each other in agreement, 0 dB = an unaltered signal. Still confused? The levels are given as a decimal fraction where 0 dB = 1.0.
It turns out that a change in volume of 1 dB approximates the just noticeable difference (jnd) in volume. Thus, sound levels are rarely stated in fractions of a dB, you just don't care. If you want to convert relative volumes (as decimal fractions) to decibels calculate the Log take the 20 * Log of the decimal to return an answer in deci (tenths) of a Bel. 0.5 or 50%, for example = 20 * Log 0.5 = 20 * -0.30 = -6 dB. It certainly sounds cooler to say "turn the vocal down by 6 dB" than to say "turn the vocal down by half" Still confused? Use your ears to set volume levels and don't exceed 0 dB on the MASTER band if the output will be rendered to wave format. My job here is done.