Hyperprism Miscellaneous Processes


Harmonic Exciter

Have you ever had an audio track that sounded dull and boring, which no amount of EQ could fix? Add a bit of excitement to your audio with Arboretum's Harmonic Exciter. Instead of simply boosting frequencies already present, as occurs with EQ, the Harmonic Exciter generates new high frequency harmonics, thereby extending the bandwidth and increasing the liveliness of the original audio.

A problem with many exciter algorithms is that the harmonic generator either does not add a significantly high level of audible harmonics or adds very unpleasant artifacts, such as harsh and gritty distortion to the signal path. With Arboretum's Harmonic Exciter great care has been taken to maximize the amount of harmonic generation, yet greatly minimize the amount of artifacts. Also, the Harmonic Exciter's phase compensated side chain avoids any phasing and frequency notching problems commonly associated with most exciter algorithms. In addition the Exciter algorithm avoids most of the aliasing artifacts that are problematic with most digitally based exciters. This is done by substantially raising the internal sampling rate of the plug-in.

Effect Usage:

The most important thing to remember when using the Harmonic Exciter is that the quality of the results are highly signal dependent. For example, certain settings may sound great on a drum track, but harsh and distorted on an acoustic guitar track. Often a fair amount of experimentation is required to achieve good results, especially because the parameters interact considerably.

A general guideline to setting the effect, is to adjust the Dynamics to around 70% and the Crossover between 2000 to 4000 Hz. For the Harmonics Type set it to "Odd and Even" for a warmer sound or "Odd" for a brighter sound. Then raise the Harmonics slider towards the maximum level, but if any unpleasant distortion is heard, then stop and back down the slider just a bit. Then adjust the Spectral Mix so the high and low frequencies are well balanced. Generally the more harmonics added the lower the Spectral Mix should be set. Then readjust the Dynamics and Crossover sliders as desired.

For something with a delicate harmonic structure such as an acoustic guitar, it may be a good idea to set the harmonics fairly low (e.g. less than 50%) and the crossover fairly high (e.g. 3000Hz). On the other hand, for audio such as a rock drum kit, it maybe more important to maximize the harmonic generation, by setting the Harmonics very high (e.g. 200%) and the Crossover fairly low (e.g. 1500Hz). Also, for speech lowering the Dynamics slider may help minimize the exaggeration of sibilance.

The Quality Level control within the Harmonic Exciter allows the user to adjust the amount of internal sample rate increase. Settings of "Good", "Better" and "Best" correspond to successively higher internal sample rates and therefore higher levels of quality. The difference in sound quality between "Good" and "Best" range from barely noticeable to fairly obvious depending upon the sound being processed. Sounds with little high frequency content may have little problem with aliasing, whereas sounds with lots of high frequency content could have lots of aliasing artifacts. Since increasing the Quality Level also greatly increases the CPU usage, it is not worth setting the Quality Level higher than needed.

For a more natural sounding effect we recommend you use the Harmonic Exciter on a relatively dry signal, and then add reverb or other effects afterwards. Although if natural is not what you're looking for, go ahead and try adding lots of reverb or other effects first. We also recommend you process each track of a mix separately, instead of processing the entire mix. The advantage of processing each track individually is that the settings can be tailored to each specific sound and inter-modulation distortion and other artifacts will generally be minimized. Certainly good results can be achieved by applying the Harmonic Exciter to an entire mix, but much more care should be taken not to over do the effect.

Parameters

Harmonics
Controls the amount of harmonics generated, with 0 to 100 percent being the nominal range. For extreme harmonic generation the harmonics slider can be set as high as 200 percent. Adjusting the Harmonics will affect the balance of the low and high frequency, so it may be necessary to adjust the Spectral Mix to compensate.
Dynamics
This novel feature adjusts the dynamics of the harmonic generation algorithm. For example, higher settings will cause the Exciter to mainly add harmonics to high amplitude transient signals only; lower settings will cause the harmonics to be applied more evenly to both low and high amplitude signals which will generally increase the overall harmonic content. The range is from 0 to 100%. Adjusting the Dynamics will sometimes affect the balance of the low and high frequency, so it may be necessary to adjust the Spectral Mix to compensate.
Crossover
Ranges from 500 to 10000Hz. Adjusts the frequency at which the harmonic generation occurs. For example, if the crossover is set to 2000Hz, then all frequencies above 2000Hz will have harmonics added.
Spectral Mix
Ranges from -24.0 to 12.0dB. Sets the gain of the frequency components which are above the crossover point.
Harmonic Type
Choices are "Odd and Even (Warm)" and "Odd (Bright)."

Quality Level

Choices are Good, Better and Best. Adjusts the accuracy of the exciter effect, by increasing the internal sample rate at which processing occurs. On the down side, raising the quality level will significantly increase processor utilization.


Bass Maximizer

The Bass Maximizer plug-in offers a wide assortment of bass enhancement and maximization effects, ranging from subtle bass enhancement to extreme bass generation. The harmonic generation feature increases the perceived amount of bass, even on low end systems with poor bass frequency response. It does so by adding harmonics which fool our brains into hearing lower frequencies than may exist in the original sound. The compression feature allows even more bass to be added without the risk of clipping or system overload.

Effect Usage:

The usage of the Bass Maximizer is relatively straightforward. A good starting point is to set the Crossover to be between 50 and 100Hz, the Low Cut to less than 20Hz, and the Resonance to less than 1.0. Then increase the Harmonics to the desired level, and add some compression. Finally adjust the Mix to balance out the low and high frequencies as desired. If deep bass is required set the Resonance to around 3.0, increase the Low Cut parameter to about 40Hz, and set the compression to around 10-20. The Bass Maximizer can be used effectively to process individual tracks, or an entire mix.

Parameters

Compression
Ranges from 0.0 to 20.0dB. Controls the amount of compression applied to the bass frequencies.
Harmonics
Controls the amount of harmonics generated, with 0 to 100 percent being the nominal range. For extreme harmonic generation the harmonics slider can be set as high as 200 percent.
Crossover
Ranges from 40 to 400Hz. Adjusts the frequency below which the harmonic generation and compression occur. For example, if the crossover is set to 100Hz, then all frequencies below 100Hz will have harmonics and compression added.
Low Cut
Ranges from 5.0 to 200.0Hz. The Low Cut feature rolls off frequency content below the specified point. This control is useful when the Bass Maximizer generates so much very low frequency content that the intended reproduction system becomes overloaded. For example, if the intended system is a typical multimedia computer which can't accurately reproduce frequencies below 70Hz, then set the Low Cut to around this frequency.
Resonance
Ranges from 0.5 to 5.0. Controls the amount of resonance applied to the low cut filter. This feature effectively adds an extra boost at the Low Cut frequency. For typical material that does not require resonance a setting of roughly 0.7 is a good starting point. For bass heavy musical styles, a setting of 3.0 to 5.0 may work best.
Mix
Ranges from 0.0 to 100.0%. Sets the mix between the frequency components which are above and below the crossover point.
Harmonic Type
Choices are "Odd and Even" and "Odd". Odd and Even gives smoother bass enhancement, while Odd harmonics will give a tighter bass sound.


Tube/Tape Saturation

Overview

The Tube/Tape Saturation plug-in has been designed to accurately model the non-linear saturation characteristics of a wide range of high-end tube amplifiers and pre-amplifiers, as well as analog reel-to-reel tape recorders. This plug-in does not model various other aspects related to tape recorders or tube amplifiers, such as high frequency roll-off or tape hiss. The Tube/Tape Saturation plug-in gives you warm, smooth tube and tape characteristics, all while staying within the digital domain.

The Tube/Tape Saturation plug-in also increases the internal processing sample rate so that aliasing artifacts are greatly reduced, something that many digitally based distortion effects neglect to do. The process of distortion adds harmonics to the signal. What happens if a harmonic is generated which has a frequency greater than the highest frequency that can be sampled (i.e. 1/2 the sample rate)? The answer is, the harmonic is folded back into audible frequency range, creating a very unpleasant type of distortion called aliasing distortion.

Effect Usage:

To produce subtle tube or tape saturation, generally adjust the Gain to between 3.0dB and 10.0dB and set the Warmth to around 80 to 90 percent. It should be noted that even a Gain of 0dB or less can often produce a considerable amount of distortion, so don't over do it. It is also generally recommended to perform the saturation before any other processing such as reverb or limiting, although certain effects such as EQ, wah wah filters, or subtle compression may work well when applied pre-saturation. But given that these are just general guidelines, certainly don't be afraid to experiment.

The Quality Level control allows the user to adjust the level of internal sample rate increase. Settings of "Good", "Better" and "Best" correspond to successively higher internal sample rates and therefore higher levels of quality. The difference in sound quality between "Good" and "Best" range from barely noticeable to fairly obvious, depending upon the sound being processed. Sounds with little high frequency content will have little problem with aliasing, whereas sounds with lots of high frequency content will have lots of aliasing artifacts. Since increasing the Quality Level also greatly increases the CPU usage, it is not worth setting the Quality Level higher than needed.

This plug-in also offers the following 4 saturation types: Tube Type1, Tube Type2, Tape Type1 and Tape Type2. (Try saying that out loud a few times...)

The tube emulations offer a significantly different harmonic structure than the tape emulations, with the tape emulation generating more higher order harmonics than the tube emulation. Tube Type1 and Tape Type1 emulate slightly more compressed tube and tape distortions, respectively, whereas Tube Type2 and Tape Type2 emulate slightly warmer, more dynamic distortions.

Another very important thing to remember when using the Tube/Tape Saturation plug-in is that because it is a non-linear effect, different results will occur by processing each track of a mix separately or processing the entire mix together. It is generally recommended that each track be processed individually. This allows the saturation settings to be optimized for each track, and generally reduces intermodulation distortion and emphasizes harmonic distortion. Certainly good results can be achieved by applying the Tube/Tape Saturation plug-in to an entire mix, but much more care should be taken not to over do the effect. Also, when applying Tube/Tape Saturation to an entire mix, it is generally best to make sure the mix has not already been limited. Not doing so will often cause ugly artifacts when the Tube/Tape Saturation is applied.

Another use for the Tube/Tape Saturation plug-in is in place of a limiter or level maximizer. Limiters or level maximizers, while generally being able to reduce the maximum amplitude with minimal distortion, often have the unpleasant artifact of squishing sound until it is dull and lifeless. On the other hand, the Tube/Tape Saturation plug-in can reduce peak level yet retain, and even add to, the punchiness of the original audio track, while often sounding warm and not overly distorted. The Tube/Tape Saturation will not work well as a level maximizer for some sounds, such as a string section, but for other sounds, such as a rock drum kit, it can be quite effective. Because Tube Type1 and Tape Type1 saturation types have higher compression characteristics, it is recommended that they be used instead of Tube Type2 and Tape Type2, when trying to perform level maximization.

Parameters

Warmth
Ranges from 0.0 to 100.0%. Controls the amount of harshness or smoothness of the saturation effect. With 0.0% tending towards the harsh side and 100.0% being smooth and warm, like a high-end tube distortion.
Gain
Ranges from -6.0 to 50.0dB. Adjusts the amount of saturation produced by the effect. For subtle tube or tape saturation stay within the 3.0 to 10.0dB range. For outrageous distortion effects try a gain of 50.0dB.
Saturation Type
Choices are Tube Type1, Tube Type2, Tape Type1 and Tape Type2.
Quality Level
Choices are Good, Better and Best. Adjusts the accuracy of the saturation effect, by increasing the internal sample rate at which processing occurs. On the down side, raising the quality level will significantly increase processor utilization.


Pitch Time Changer

The Pitch Time Changer lets you vary the playback speed (duration) and the pitch of the input signal independently.

For pitch-shifting, the algorithm works by loading memory with an incoming signal sampled at rate A and reading out the samples at rate B. The ratio A/B determines the pitch change. To maintain a continuous output signal, samples must be repeated (for upward pitch shifts) or skipped (for downward pitch shifts). Because the output address pointer repeatedly overtakes the input address pointer (for pitch increases) or is overtaken by the recirculating input address pointer (for pitch decreases), the output address must occasionally jump to a new point in the memory. To stretch the time base of the input signal, the algorithm repeats small "grains" (segments) of the input signal, while for time shrinking, it deletes intermediate grains.The sound quality of the Pitch Time Changer is based on the nature of the input signal and on the ratio of pitch change it is asked to perform. Small pitch and time changes tend to generate less-audible side effects.

There are no "magic" settings. Only by experimenting with a given input sound can you explore the range of sound transformation possibilities this effect can realize.

Hyperprism 2 users: Pitch Time Changer's time calculations, and the resultant sound quality, are influenced by the Processing Preferences settings found in HyperEngine's Preference menu. Try boosting the filter and interpolation settings if your time slow-downs are sounding too grainy.

Do not be afraid to experiment with what might be considered "absurd" settings. For example, extremely slow playback rates can lead to very interesting and un-recognizable sounds.

At some extreme settings of the Pitch Time Changer, especially when very high feedback settings are in use, the algorithm may mute its output when certain processing limits are exceeded. In this case, simply move the mouse back to a slightly less "extreme" setting and sound playback will resume.

Hyperprism 2 users: Pitch Time Changer won't do time changes on live signal when used with a HyperEngine Play-Thru document. This process is meant more for use on disk files, which are processed in HyperEngine File documents.

Parameters

Speed
Speed variation, where 1% results in an unrecognizably-elongated version of the sound and 200 corresponds to a double-speed rendition of the input signal. Settings up to 400% (4x normal speed) are permitted.
Pitch
Pitch variation from -200% to +200%, where 0% represents a transposition towards extreme low frequencies and 200 transposes the input sound up an octave.

Note that negative values of pitch shifting cause the pitch of the source signal to be shifted the same amount as positive percentage values, but the playback direction of the individual sound fragments, or "grains" are reversed. This can be especially interesting on speech or rhythmic sounds. For example, a pitch shift of -100% results in playing the sound at it's original pitch, but broken into small, backwards chunks.

Mouse click in the numeric field to the left of the Pitch slider, and drag the low-range value down to -200%. Now you can experiment with negative pitch shifts.

Grain Size
To stretch or shrink the time base of the original sound signal, the algorithm repeats (or skips) small segments, or "grains" of the original sound. The length of these grains can be varied, resulting in a great difference in sound. The effect can range from echoing to "granulation" to obliteration of the identity of the input signal.

Small grain settings can lead to buzzing sounds, while long ones may be perceived as a doubling, or echo effect. The actual results depend heavily on the character of the original sound. Hyperprism works with a maximum grain size of 1024 samples; this parameter is controlled by a percentage value, ranging from 0 to 100%.

X-Fade Size
The X-Fade slider sets the crossfade time, from 0% (no crossfade) to 100% (where the crossfade lasts the entire length of the grain. Short settings tend to produce audible artifacts at the boundaries set by the Grain size.

X-Fades of zero will often click. Longer crossfades will be smoother, but can lead to more chorusing or doubling sounds.

Feedback
The Feedback slider introduces regeneration, where the output of the pitch time changer is mixed back at it's input. A setting of 0% is no feedback, 100% will lead to run-away regeneration, as the full output of the effect is constantly being routed to it's own input.

Different feedback settings can result in anything from flanging or resonant effects to cascading showers of pitch-bending echoes as the mouse is moved.

Pitch Change Mix
Ranges from 0-100%. Sets the mix of time stretched sound and pitch shifted sound. 0% settings allow you to hear only time stretched signal; 100% means all pitch shifted signal.


Pitch Changer

This effect is called "Pitch Shift" in Hyperprism-MMP. The Pitch Changer lets you vary the pitch of the input signal.The algorithm works by loading memory with an incoming signal sampled at rate A and reading out the samples at rate B. The ratio A/B determines the pitch change. To maintain a continuous output signal, samples must be repeated (for upward pitch shifts) or skipped (for downward pitch shifts). Because the output address pointer repeatedly overtakes the input address pointer (for pitch increases) or is overtaken by the recirculating input address pointer (for pitch decreases), the output address must occasionally jump to a new point in the memory.

There are no "magic" settings. Only by experimenting with a given input sound can you explore the range of sound transformation possibilities this effect can realize.

At some extreme settings of the Pitch Time Changer, especially when very high feedback settings are in use, the algorithm may mute its output when certain processing limits are exceeded. In this case, simply move the mouse back to a slightly less "extreme" setting and sound playback will resume.

Parameters

Pitch
Pitch variation, from -200% to +200%, where 0% represents a transposition towards extreme low frequencies and 200 transposes the input sound up an octave.

Note that negative values of pitch shifting cause the pitch of the source signal to be shifted the same amount as positive percentage values, but the playback direction of the individual sound fragments, or "grains" are reversed. This can be especially interesting on speech or rhythmic sounds. For example, a pitch shift of -100% results in playing the sound at it's original pitch, but broken into small, backwards chunks.

Mouse click in the numeric field to the left of the Pitch slider, and drag the low-range value down to -200%. Now you can experiment with negative pitch shifts.

Grain Size
To stretch or shrink the time base of the original sound signal, the algorithm repeats (or skips) small segments, or "grains" of the original sound. The length of these grains can be varied, resulting in a great difference in sound. The effect can range from echoing to "granulation" to obliteration of the identity of the input signal.

Small grain settings can lead to buzzing sounds, while long ones may be perceived as a doubling, or echo effect. The actual results depend heavily on the character of the original sound. Hyperprism works with a maximum grain size of 1024 samples; this parameter is controlled by a percentage value, ranging from 0 to 100%.

X-Fade Size
The X-Fade slider sets the crossfade time, from 0% (no crossfade) to 100% (where the crossfade lasts the entire length of the grain. Short settings tend to produce audible artifacts at the boundaries set by the Grain size.

X-Fades of zero will often click. Longer crossfades will be smoother, but can lead to more chorusing or doubling sounds.

Feedback
The Feedback slider introduces regeneration, where the output of the pitch time changer is mixed back at it's input. A setting of 0% is no feedback, 100% will lead to run-away regeneration, as the full output of the effect is constantly being routed to it's own input.

Different feedback settings can result in anything from flanging or resonant effects to cascading showers of pitch-bending echoes as the mouse is moved.

Mix
The Mix parameter varies the balance between the level of the original signal and the new, pitch-shifted signal.


Vari-Speed

Emulates the behavior of analog recording devices; as you slow down the playback speed, the pitch is also lowered. Speed up the playback and the pitch is also raised. With adjustments ranging from twice as fast to 100 times slower, you can create sub-bass growls or rapid-fire speech.

Hyperprism 2 users: Vari-Speed's time calculations, and the resultant sound quality, are influenced by the Processing Preferences settings found in HyperEngine's Preference menu. Try boosting the filter and interpolation settings if your time slow-downs are sounding too grainy.

Vari-Speed won't do time changes on live signal when used with a HyperEngine Play-Thru document. This process is meant more for use on disk files, which are processed in HyperEngine File documents.

Parameters

Playback Speed
Control the playback speed of the input sound, from 1/100th of original to 400% of original speed.
LFO Frequency
Modulates the playback speed to create subtle shifts or huge swings in speed, from 0 to 20 Hz. Note that when previewing in real-time, frequencies above 5-10 Hz may sound similar depending on the speed of your machine. Try processing faster LFO frequencies to a new file if you have this problem.
LFO Depth
Controls the degree of shift in playback speed induced by the LFO. Note that with 100% depth the playback speed may shift to very slow depending on the ranges you have set for the Playback Speed parameter.


Noise Gate

The Noise Gate provides a way to remove unwanted quiet portions of a sound. The most important parameter of the noise gate is the threshold. When the level of your audio is softer than the volume set by the threshold, the Noise Gate effect will turn the audio off, leaving silence. However, when the signal level exceeds the threshold, the audio will be automatically turned back on. Setting the threshold to "0" will result in nothing but silence, since the audio can never get above that point. Similarly, setting the threshold to a very low number (the minimum is -96 dB) will result in a gate that never "closes," in other words, the sound will never be shut off.

The Noise Gate is "stereo coupled" when processing a stereo file. Left channel and right channel signal levels are calculated jointly and one gain is applied to both channels. This prevents weird shifts in the stereo balance.

Parameters

Attack
This parameter is called "Open Time" in Hyperprism-VST. It controls attack time from 0 milliseconds to one second. When the gate is closed (meaning the Noise Gate is shutting off your sound), it is constantly watching for the signal to exceed the level of the threshold. When this happens, the gate will open and begin passing sound again. However, the gate can be programmed to open immediately, or to "ramp" open at a particular rate. This rate is set by the attack time parameter. For example, if you are using the noise gate on a percussion track, set the attack time very short, unless you want to create a "fade in" effect on each drum transient. Feel free to experiment!
Threshold
Threshold, from -96 to 0 dB. The threshold sets the level at which the Noise Gate opens and closes. Audio is passed through the Noise Gate when it's signal level exceeds the Threshold setting. When the level of the signal drops below Threshold, it is muted. The rate at which the gate opens and closes is set by the Attack and Release times.
Release Time
This parameter is called "Close Time" in Hyperprism-VST. The Release Time is a complement to the Attack time parameter (described above). When the signal falls below the Threshold, the gate closes. This can happen instantaneously, or can be programmed to fade out more slowly. The Release time control sets the duration of this fade out. If you want to try and preserve a sense of room ambience when the gate closes, set the release time to a longer value. Or, if you are trying to create a very abrupt truncation of a sound, use a fast Release time. This parameter ranges from 0 to 1000 ms.
Metering lights

Red light indicates signal is below the threshold, so gating is applied. Green light means signal exceeds threshold, no gating is applied and signal passes unchanged. Metering lights are not included in Hyperprism-DAS for AudioSuite.


Compressor

The Compressor effect lets you manipulate the dynamic range of an audio signal. It does this by reducing the peak levels of the signal and then bringing up the overall level by a corresponding amount. The result is that "soft" sounds seem "louder," and the overall signal's dynamic extremes are smoothed out.

The Compressor is "stereo coupled," left channel and right channel signal levels are calculated jointly and one gain is applied to both channels. This prevents abnormal shifts in the stereo balance.

The Hyperprism Compressor features new discrete Ratio and Threshold settings, new Hard Knee/Soft Knee option and smoother, more transparent sound.

Parameters

Volume (or "Output Gain")
This control allows you to vary the overall output volume of your signal after it has been compressed. This parameter is called Output Gain in Hyperprism 2, Hyperprism-MMP and Hyperprism-DAS and ranges from -20 to 20 dB in the Hyperprism 2 and Hyperprism-MMP editions. Volume ranges from -36 dB to +36 dB in Hyperprism-VST and Hyperprism-DAS.
Attack
Attack time from 0 ms to 1000 ms. As the compressor operates, it is constantly watching the level of your audio signal and adjusting it's dynamic range. The Attack parameter determines how quickly the compressor will respond to the signal's changing dynamics. Slower Attack times will let individual transients, such as drum hits, through without turning them down. Faster attack times will "catch" these transient events.
Release
Release time from 0 ms to 1000 ms. As the compressor operates, it is constantly watching the level of your audio signal and adjusting it's dynamic range. The Release parameter determines how quickly the compressor will respond to the signal's changing dynamics. Slower release times will generally result in less noticeable artifacts in your signal's low-level sounds, such as room ambience. Fast release times may result in the room sound fluctuating in level, perhaps in a distracting way. On the other hand, this might be just the effect you are looking for. Experiment!
Ratio
As in any traditional hardware compressor, Hyperprism offers a Ratio setting to control the amount of gain reduction. A ratio of 1:1 means no reduction, and a ratio of 10:1 means a 10 dB increase in the input signal results in only 1 dB increase in the output signal. Possible settings range from 1:1 to 10000:1. In Hyperprism-MMP this value is called "Compression Ratio."
Threshold
This control allow you to set the volume level above which compression begins. Possible values range from -60 to +20 dB. Use the metering LED's to note when your input signal is moving above and below the threshold. In Hyperprism-MMP this value ranges from -60 to 0dB.
Soft/Hard Knee
When in hard knee mode, gain is applied as soon as the input signal crosses above the threshold (after attack and release times, if any, are considered). The soft knee gradually applies the compression gain to the input when the signal is from -6 dB below the threshold to 6 dB above the threshold. Above 6 dB over the threshold, full compression gain is applied. Using the soft knee allows for more compression with smaller attack and release times.

Output characteristic charts showing the difference between Hard Knee and Soft Knee compression

Metering lights
Green light indicates signal is below the threshold, no processing is occurring. Orange light denotes signal in the soft knee region and incremental compression is occurring. Red light means signal above the threshold, full compression value is being applied. These are set taking into account the attack and release times, so that they light up only when the gain is actually being applied. Metering lights are not included in Hyperprism-DAS for AudioSuite.

Limiter:

Like the Compressor, the Limiter effect lets you manipulate the dynamic range of an audio signal. It does this by sensing the peak levels of the signal and attenuating those peaks by adding an adjustable amount of gain. Reducing the peaks allows you to bring up the softer parts, making your file sound louder.

As signal rises toward zero dB, no limiting is applied. Any signal that exceeds zero dB will be limited to that setting (when Threshold is set to zero dB).

Set the threshold for the level at which you want the limiter to kick in. As long as your signal is close to, or slightly above the threshold, the output level of the signal will match your gain setting. When you put the threshold below the level of incoming signal, you'll be attenuating signal which exceeds the threshold, resulting in a perceived increase in the loudness of the signal. If the threshold is set above the level of incoming signal, no processing is applied.

The limiter is "stereo coupled," meaning the signal level is calculated by considering the left and right channels at the same time; one gain setting is applied to both channels. This prevents undesirable shifts in the stereo balance.

You may notice that the Hyperprism Limiter has a Release time, but no Attack time setting; we've set attack time parameter permanently at 0, so it responds immediately to increases in the input signal.

Ceiling
Threshold setting, from -96 to 0 dB. Input signal exceeding threshold will have the volume setting applied to it.
Volume
The amount of gain applied to the signal, ranging from -36 to 36 dB.
Release
An adjustable time lag, you determine how long it takes for limiting to cease once signal falls below threshold. Measured in milliseconds from 0 to 1000 (one second).
Metering lights
The green light means signal is below threshold and no limiting is occurring. The red light indicates that input has exceed the threshold and limiting is engaged. Metering lights are not included in Hyperprism-DAS for AudioSuite.

Sonic Decimator (TM)

The Sonic Decimator allows you to reduce the apparent sample rate or bit-depth of your sound file. This is an excellent way to re-create the sound of the first generation of digital sampling devices. At moderate settings, the can create a "low-fi" effect, as though the sound was being processed by a low-budget sampler. At extreme settings, large amounts of hiss and distortion, as well as intense aliasing effects are added to the sound.

Parameters

Sample Rate
This control reduces the apparent sample rate of your file. This has the effect of reducing the sound's high frequency content and can introduce intense aliasing at very low settings. Default range is 1% to 100%.
Bit Depth
This parameter reduces the files equivalent-bit-depth, reducing the overall dynamic range, increasing the noise floor and adding digital resolution distortion. Default range is 1 bit to 16 bits.


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