Dynamics

About Dynamic Processing

A dynamic processor is used to alter the dynamic properties of the recording. To understand how a dynamic processor works, imagine a sound engineer trying to maintain as steady a volume level as possible during a recording. When the input level increases he pulls down the volume fader, and he pushes it up when the input level decreases. A dynamic processor does the same thing automatically according to its settings, only with a much faster reaction time.

Modern dynamic processors allow you to set a ratio between the input levels and the output levels. This ratio is visualized as a curve where the horizontal axis represents the input level and the vertical axis represents the output level. A straight line as shown below represents a 1:1 ratio.

With such a setting, no change is made to the level as it is processed. Changes are made to the dynamics by altering the ratio and the threshold. In the example below, all signal levels above -20 dB are attenuated with a 100:1 ratio, so that the output level only raises 1dB when the input raises 100dB above the threshold. This setting would be comparable to a piece of hardware known as a limiter. You can see from the graph below that once the input level reaches –20 dB, the output level is barely going any higher than -20 dB, even when the input level increases.

If the dynamic processor changes the level too fast, low frequency signal components might become distorted. How quickly the dynamic processor adapts to changes in the input level is called response time. The response time is divided into the amount of time when the input level rises (the attack time) and when it falls (the release time).

When applying compressor ratios that lead to extreme changes in the dynamics, audible artifacts might become noticeable (often referred to as "pumping and breathing"). Smoother compression curves will generally reduce the artifacts of a dynamic processor. Soft kneeing automatically softens the curve to reduce such artifacts. A high level of soft kneeing was used in the image below.

User Interface

The main UI of the Dynamics processor is shown below. The controls are discussed in more detail in the Parameter Settings section below.

Parameter Settings

• Threshold level (-60 dB to 0 dB)

When the input level exceeds the set threshold, the compressor starts to apply compression. For example, if the threshold is set at -12 dB, it will only respond to incoming audio that exceeds -12 dB.

Note that as soon as you activate the Soft knee, it's possible that compression takes place even before the input level exceeds the set threshold, due to the nature of the soft knee.

• Ratio (0.01:1 to 100:1)

By setting up a ratio, you decide how much gain reduction should take place for every dB the input level exceeds above the set threshold. For example, if the threshold is set on 2.00:1, there will only be a 1 dB output increase for every 2 dB the input level exceeds above the set threshold. You could also say that the compressor will do 1 dB of gain reduction for every 2 dB the input level exceeds the set threshold.

It's also possible to set a ratio that is below 1.00:1. This basically turns the compressor into a gate or expander. Audio levels below the threshold will now be reduced.

• Range (0 to 60 dB)

The range control sets the maximum gain reduction of the dynamic processor.

• Soft knee (0 dB to 24 dB)

A soft knee allows you to create a smoother compression curve, which can help to prevent or reduce audible artifacts. With the soft knee set to 0 dB, there is no soft knee applied to the compression curve. When the soft knee is set to 8 dB for example, the compression curve will be smoothly interpolated for input levels in the range -8 dB to +8 dB relative to the threshold level.

• Attack time (0.01 ms to 200 ms)

You can use the attack time knob to adjust how quickly the compressor will react to incoming audio that is above the threshold. The faster the attack time, the faster the compressor will react.

If compression is applied on a snare for example which has a very loud attack, setting up a very fast attack time might help to reduce the loud attack of the snare, as the compressor can almost instantly apply gain reduction. However, in many scenarios a slightly slower attack time might provide you with a more natural sounding compression behavior, for example during mastering.

• Release time (ms)

You can use the release time knob to adjust how quickly the compressor will stop processing audio that is no longer above the threshold. A fast release time can greatly enhance the sustain/decay of certain softer elements of the audio, but it can also introduce a "pumping" effect. Slow release times (>200ms) often result in a more smoother/natural compression behavior.

• Auto-Release button

You can enable auto-release mode by clicking the button labeled "A" on the left side of the numerical entry field for the release time. In auto-release mode, the release time specifies the maximum release time and the dynamic processor shortens the release time for sharp transients.

• Hold time (0 ms to 500 ms)

As soon as audio is below the set threshold, the release time kicks in. However, the hold time allows the compressor to postpone the activation of the release time for up to 500ms, which might provide you with a smoother compression behavior in certain audio material.

Although you are free to experiment with the hold function, we would like to suggest to use a hold time that is significantly shorter than the release time.

• Make-up (0 dB to 32 dB)

You can use the make-up knob to compensate for the level reduction caused by the compressor. It's possible to set the make-up in auto mode, by pressing the A button in front of the knob. When the A button is yellow, the auto make-up is activated and the make-up knob is disabled.

Since compression reduces level, make-up gain is useful to restore the audio to the same perceived volume as before compression is applied. When auto mode is enabled, Acoustica will automatically apply make-up gain to the output to match the input volume.

• Latency (0 ms to 30 ms)

It is possible to reduce harmonic distortion of low frequency content even with short attack and release times if the dynamic processor is allowed to examine the signal ahead of time. The downside is a slightly increased latency. You can adjust the maximum allowed latency in milliseconds.

• Channel Linking (0% to 100%)

With the channel linking set to 100%, the amount of gain reduction will be the same for the left and right channels, even if there is a difference in level between these channels. The more you dial the knob away from 100%, the more the channels will be independently treated, with complete independence between the channels once you have a 0% channel linking.

Channel linking values below 100% can result in shifts in the stereo image. During mastering, the best results will be most likely achieved with the channel linking set between 70% and 100%.

• Mix (0% to 100%)

The mix parameter controls the unprocessed and processed signal mix. Normally, this will be set to 100%, while lower settings allow for convenient parallel compression. Parallel compression can work well with heavier compression settings to restore a more natural sound.

• Oversampling (off, 2x, 4x)

The internal sample rate of the dynamic processor can be multiplied by 2 or 4, depending on the sample rate of the project. For example, if you have a project with a sample rate of 48 kHz, setting the oversampling to 4x will result in an internal sample rate of 192 kHz.

Side Chain Filtering

The signal that is used to calculate the gain signal in dynamic processing is referred to as the side chain signal. You can filter the side chain signal in Dynamics so that the dynamic processing is more sensitive in certain frequency regions. This works similar to an equalizer.

Click the button labeled Side chain filter in the bottom left corner of the Dynamics user interface to open (or close) the side chain filtering settings:

The side chain filter in Dynamics allows you to filter the side chain signal similar to the way an equalizer works.

You can click one of the filter buttons to toggle the filter and the following filter types are available:

: Low cut, removes frequency content below the band frequency

: Low shelf, boosts or attenuates frequency content below the band frequency

: Peak filter, boosts or attenuates frequency content around the band frequency

: High shelf, boosts or attenuates frequency content above the band frequency

: High cut, removes frequency content above the band frequency

The Side chain filter button turns orange as soon as filtering is active. In the above image, a low cut and bell filter have been activated on the side chain.