Peak and RMS values are a hot topic in the world of audio and so it comes to no surprise that these different values also apply to dynamic range compression.
What is peak compression? Peak compression means that the compressor will react according to the peak of the input or sidechain signal. As the peak of the compressor control exceeds the threshold, the compressor will kick in.
What is RMS compression? RMS (root mean square) compression means that the compressor will react according to the “average loudness” of the input or sidechain signal. As the “average” of the compressor control exceeds the threshold, the compressor will kick in.
In this article, we’ll discuss the difference between peak and root mean square values and how compressors will use these values in order to compress the dynamic range of their input signals.
A Primer On Audio Signals
To truly understand dynamic range compression (and, therefore, both RMS and peak-metering compression), we must understand the audio signals that are subjected to compression. In this primer section, we’ll learn the basics of audio signals.
Audio signals can be simply defined as electrical energy (active or potential) that represents sound.
The waveforms that make up audible sound waves are effectively mimicked in an audio signal. These waveforms have peaks and troughs of maximum and minimum sound pressure levels. Sound waves effectively disrupt the molecules of a medium as they travel through the medium.
The universally accepted range of human hearing is between 20 Hz and 20,000 Hz.
Audio signals, then, are largely defined within this same 20 Hz – 20 kHz limit (though signals may have frequencies beyond this band).
Analog audio signals are defined as AC (alternating current) signals with an amplitude defined by the voltage levels of the signal. These amplitudes may be given in volts (V) or millivolts (mV). They may also be given as decibels relative to 0 volts (dBV) or decibels relative to 0.775 volts (dBu).
An analog audio signal can be represented by the following 1 kHz sine wave:
The amplitudes of analog audio signals vary greatly depending on the device. For example, microphones can only handle relatively small “mic level signals” while speakers require much largely signals to drive their drivers. The different nominal signal levels are listed in the following table:
|Signal Level Type||Typical Nominal Signal Level|
|Mic Level||-60 dBV (1 mVRMS) to
-20 dBV (100 mVRMS)
|Line Level (Consumer)||-10 dBV (316 mVRMS)|
|Line Level (Professional)||+4 dBu (1.228 VRMS)|
|Instrument Level||-20 dBu (77.5 mVRMS)|
|Speaker Level||20 dBV (10 VRMS) to
40 dBV (100 VRMS)
Mixing and mastering (along with audio storage) will practically always be a line level (professional line level, ideally). Therefore, when dealing with compressors, the input signals (and output signals) will generally be around the +4dBu level.
Digital signals are discrete-time representations of their continuous-time analog counterparts and have amplitudes/levels defined relative to a maximum amplitude at 0 decibels full-scale (0 dBFS). 0 dBFS is the absolute maximum amplitude, based on bit-depth, that a digital audio signal may have before digital clipping occurs.
The aforementioned 1 kHz analog sine wave, in digital form, can be visualized with the following image (the digital resolution is defined by a 48 kHz sample rate and 24 bit bit-depth):
In this article, we’ll be concerned with the amplitude of audio signals.
To recap this primer:
- Audio signals are electronic representations (potential or active electrical energy) of sound waves.
- Audio signals can be stored and played back in analog (continuous-time) or digital (discrete-time) formats.
- Audio signals are typically defined within the frequency range of 20 Hz to 20,000 Hz but can have information beyond this band.
- Analog audio signal amplitude is typically defined as an AC voltage, dBV or dBu.
- Digital audio signal amplitude is typically defined relative to the maximum digital amplitude before clipping (0 dBFS).
- Compressors (and other audio processors/effects) will generally act upon line level signals.
This explanation, again, is very brief. For more in-depth information, please consider reading the following My New Microphone articles:
• What Is The Difference Between Sound And Audio?
• What Are Decibels? The Ultimate dB Guide For Audio & Sound
Now that we understand that audio signals will have varying amplitudes, we can understand compression.
The whole idea of compression is to decrease or “compress” the dynamic range of an audio signal. The dynamic range is defined as the range between the highest amplitude (the “loudest” part of the audio signal) and the lowest amplitude (the “quietest” part of the audio signal).
Peak Vs. RMS Values
Now that we understand how audio signals are defined by their amplitudes and how compressors act to compress the dynamic range of audio signals, we’ll move on to our discussion on peak and RMS values.
When discussing audio, peak and RMS will generally always be in reference to a signal level. It could refer to the power, voltage, dBFS or other units that define the “strength” of a signal or a device’s ability to handle the “strength” of a signal.
In the case of compressors, these terms apply to how the compressor will meter its sidechain signal.
Peak amplitude refers to the absolute highest instantaneous voltage level of the signal. This can be visualized in the following image (I’ll continue using sine waves for simplicity):
For simple sine waves, we have the following equation to calculate the signal’s RMS level from its peak level.
What we end up with it VRMS = 0.707 VPeak.
For more complex audio signals, we technically have the following, rather complicated, equation:
Essentially, though, RMS-level audio is a measurement of a bipolar AC signal’s average signal strength. Bipolar simply means that the signal has both positive and negative amplitudes throughout its waveform.
Peak Vs. RMS Compression
Now let’s move onto compressors. A compressor may be designed with peak and/or RMS metering. Note that only one metering system can be used at once.
Each and every compressor has a two main components:
- A gain reduction circuit/element that compresses the input/program audio.
- A sidechain path that produces a control signal (from the program signal or an external signal) that will control how the gain reduction circuit/element attenuates the input/program audio.
This control signal (also referred to as the sidechain) is derived from the input audio signal (common) or via an external audio signal (less common). It can be further manipulated to achieve the typical compressor parameters (threshold, ratio, attack time, release time, knee, and/or lookahead).
To learn more about these compressor parameters, check out the following My New Microphone articles:
• Dynamic Range Compression: What Is The Threshold Control?
• Dynamic Range Compression: What Is The Ratio Control?
• Dynamic Range Compression: Attack & Release Controls
• Dynamic Range Compression: What Is The Knee Control?
• Dynamic Range Compression: What Is The Lookahead Control?
The selection of the sidechain signal can be visualized with the following signal flow diagram of a feedback compressor:
Regardless of what the sidechain signal is, it must pass through a level detection circuit in order to converted to a control signal for the gain reduction circuit. More specifically, the sidechain audio signal (AC) must be rectified and turned into a DC control voltage for the GR circuit.
This is where peak and RMS metering comes into play.
A peak compressor’s level detection circuit will measure the peak level of the incoming AC sidechain signal and effectively output a varying DC voltage equal to the peak level of the incoming signal.
An RMS compressor’s level detection circuit, conversely, will measure the RMS level of the incoming AC sidechain signal and effectively output a varying DC voltage equal to the RMS level of the incoming signal.
Fortunately, the RMS of an AC signal is equal to the level of the DC signal that would provide the same average power so it’s relatively easy to detect/rectify RMS levels.
The exact components and layout that will make up either level detection circuit will vary from compressor to compressor.
Due to the nature of RMS and peak levels, RMS and peak compressors will react differently.
The most obvious difference is that peak detectors will produce control signals with higher amplitudes. In terms of triggering the compressor, a peak compressor will require a higher threshold than an RMS to achieve the same trigger point when prompted with the same sidechain audio.
RMS detection is also more gradual and will not react so abruptly to the changes in incoming signal level.
The window of an RMS level detector refers to the length of time the detector will hold in its memory to measure the “average” of the signal. Shorter windows make the RMS detector more reactive and longer windows take the average over a longer time horizon.
Note that window size is not the same as the compressor attack and release times. Note, too, that a theoretical window size of zero would effectively turn the RMS detector into a peak detector.
Peak detectors can be used for limiting, where fast reaction times are required to clamp down on any and all audio that surpasses the threshold.
RMS compression is often more subtle and longer lasting (given the same attack/release times) than peak compression, making it a superb choice for non-percussive instruments, singing and mix buses.
Because it only acts upon the average amplitude, RMS compression can potentially miss transient peaks in a signal completely.
Peak compression is often more noticeable, making it useful on percussive/transient-heavy signals; for pumping effects, and, in the right context, sidechain compression.
Examples Of Compressors With Peak & RMS Options
Before we wrap things up, it’s always a great idea to consider some examples of compressors that offer both peak and RMS metering options.
Note that I’ll link specific terms in this section to in-depth My New Microphone articles on the subjects.
Rupert Neve Designs Portico II
The Rupert Neve Designs Portico II Master Buss Compressor (link to check the price at B&H Photo/Video) is a stereo VCA compressor. Each of the Channels (A & B) have a pushbutton to select between RMS and peak metering.
As its name suggests, the Portico II works tremendously well as a stereo master buss compressor. It’s beautifully transparent and fully capable of adding weight, width and density; subtle harmonics saturation/grit; low-end emphasis; mixing/mastering “glue”; limiting, and more to the output bus of a mix (or another track/bus if necessary).
In addition to the Peak/RMS metering option, this highly versatile compressor features the standard attack, release, threshold, ratio and makeup gain control. It also has a feedback/feedforward circuitry option for a more relaxed or harder/faster compression style, respectively.
For more information on feedback and feed-forward compression, check out my article Feedback Vs. Feed-Forward Dynamic Range Compressors In Audio.
Each channel also has a high-pass filter; a blend knob for parallel processing, and two flavours of harmonic distortion via the customary silk circuit with a dedicated texture control knob. Each channel can also be side-chained if need be.
Finally, mid-side tweaking is made possible with the Stereo Field Editor, which allows users to shape the stereo spread of program material. Bring out the sides with the width control and the mids with the depth control. Both the width and depth controls have their own low, low-mid, high-mid, or high options for a targeted response in the frequency spectrum.
Rupert Neve Designs is featured in the following My New Microphone articles:
• Top Best Studio Recording/Mixing Console Brands
• Top Best Audio Compressor Brands In The World
• Top Best Audio Equalizer Brands In The World
• Top Best Audio Brands For 500 Series Modules/Equipment
The MeldaProduction MCompressor (link to check it out at Plugin Boutique) is an awesome free compressor plugin that offers a range of RMS window sizes (from 0 to 100 ms) along with peak metering.
In addition to all the standard controls mentioned previously (attack, release, threshold, ratio, makeup gain), the MCompressor also has the choice of three knee controls and the ability to customize the shape of the compressor’s input/output relationship graph/curve.
By shaping this curve, users can effectively turn the MCompressor into an expander, gate, “rhythmizer”, limiter, or upward compressor.
This plugin also offers a low pass filter along with a high pass filter. On top of that, there’s a signal maximization option to bring the maximum signal up to 0 dB.
This compressor can be side-chained and can process up to 8 channels of audio at once.
Should you EQ or compress first? There is great debate as to whether an equalizer or a compressor should come first in the audio signal chain. There’s no rule stating that either should come first. However, in general, you’ll likely get the most out of the EQ and compressor if you follow these standards:
- For tonal shaping, it’s often best to compress beforehand to avoid having to alter the compressor settings.
- For audio signals with that require significant filtering, it’s best to EQ first so as to not feed the compressor with unwanted frequency content.
Should you compress every track in a mix? As a general rule, compression should be used with intent and, therefore, only be used on every track in the case that every track would require it. More often than not, there will be certain tracks in a mix that sound perfectly fine (and better) without dynamic range compression.
Once again, the typical benefits of using compression on a track include (but are not limited to) the following:
- Maintaining a more consistent level across the entirety of the audio signal/track
- Preventing overloading/clipping
- Sidechaining elements together
- Enhancing sustain
- Enhancing transients
- Adding “movement” to a signal
- Adding depth to a mix
- Uncovering nuanced information in an audio signal
- “Gluing” a mix together (making it more cohesive)