DSP (Native Effects)
Related topics
DSP-chain in general
Controlling meta/dsp/vst devices using Automation
Controlling meta/dsp/vst devices using pattern effect-commands
DSP Devices:
Pre mixer
Bus Compressor
Maximizer
Chorus
Gainer
DC Offset
Stereo Expander
Gate 2
Filter 2
EQ5
EQ10
Mixer EQ
Compressor
Distortion 2
LoFi Mat
Delay
Phaser
Flanger
Reverb
MP Reverb 2
Renoise currently counts 19 native effects (above the primary panel) that you can apply to your tracks, your send-tracks or the mastertrack.
The old effects from 1.8 will load and show when loading them from 1.8 songs. Consult the 1.8 manual for the older DSP effects (Filter, Gate, Stereo expander, mpReverb etc.)
Let’s deal with them one by one…

This is the primary effect-panel, in here you arrange the global track-volume, panning and width for your current track. This one cannot be removed or replaced somewhere else in the rack.
- Active - Indicates track is currently active (activate with “\”)
- Off - Indicates track is being turned off (deactivate with “\”)
- Mute - Mute the current track (Activate / mute with “lshift-\”)
- Solo - Solo the current track (disable the rest) (“lctrl-\”)
- Unmute all - Enables all tracks
- Track-name - The Custom Trackname caption is user customisable, this can be done on top of the track itself or here in the master DSP.
- Routing - Sets Channel Routing (ASIO only) to the available outputs on your audio-device.
The amount and type of outputs is vendor dependant. - Panning - Sets default panning for the current track
- Volume - Sets default volume for current track
- Width - Sets default width for current track
Alternating effects
Effects that alter sound-output are usually placed to spice up the output of a track or to give it a bit of a boost or just vice versa, crippling it or pushing it more to the background.

A dynamics processor that softens signals (lowers its volume) above a specified amplitude level. The bus compressor, in contrary to the “normal” compressor, analyzes the incoming signals like feed-forward /and/ feed-back compressors. Short peaks are handled via a feed-back algorithm, constant signals via a feed-forward algorithm. This way both compression methods are used where they work optimal.
The bus compressor is a perfect tool for mastering or levelling. Use the “normal” compressor if you want to really punch/squeeze your sound - fatten your drums.
- Threshold - The dB value where compression kicks in
- Ratio - Threshold factor appliance. If threshold is exceeded, compression ratio is being applied to the exceeding peak.
- Attack - The attack setting determines how fast the compressor recognizes the incoming signal “peaks”
- Release - Release sets how long it takes for the compressor to come back from a “peaked” state
- Makeup - Final gainer that is applied behind the compressors output
- Knee - dB offset base from the bendpoint of the threshold factor

The maximizer allows you to boost and limit audio. It will hardclip signals that exeeds the threshold, but then soften the release of the signal when it falls below the threshold (in contrary to plain hard clipping).
Use the maximizer to punch or squeeze sounds, or as final limiter on the master track to avoid rough hard clipping.
- Boost - Gainer, that is applied before the threshold is measured
- Threshold - The dB value the maximizer limits the input to
- Peak Rel. - Compression release factor for peaks (transients)
- Slow Rel. - Compression release factor for non peaks
- Ceilling - Gainer that is applied after the maximizer limited the signal
The small red softclip led indicates wether the peak (transient) release factor or slow release factor is used.

The chorus effect enriches your audio by adding warmth and dimension to the signal.
- Rate - Amount applied.
- Depth - Depth applied.
- Feedback - Amount of feedback injected into the signal.
- Delay - amount of delay added to the signal.
- Dry/Wet - Mix dry / wet signal output.
- Phase - Angle of shift.
- Filter type selection - Select type of filter to apply upon effect signal.
- Filter Freq. - Frequency range to apply filter to.
- Filter Reso. - Resonance amount generated within the effect signal.

The gainer turns up the volume above the current volume setting of the primary panel. This one can be used to turn up volume of samples that sounds too silent for usage and you can’t maximize the sample any further in the sample editor. The panning value can be used to set a base for the global panning. Slide the primary panel-panning slider all the way to the left or to the right, then play with the gainer panning to listen what the effect is. The inverse buttons inverse the audio’s channel.

The DC-offset panel sets the wave xx% above or below the sample it’s dc-line (or flatline which means a total silent volume amplitude). This panel can be used to change the sample-level and prevent peaks of the samples causing clicks or hiss-points when peak-rates touches the max-level (or crosses that line). Samples that require this correction are mostly sampled with a bad amplitude or resonance. In the sample editor you can correct this hardwired with the DC-adjust button. Beware that improper use may cause damage to your (old) speakers as they may not be capable of handling impulses by clicks and ticks. There may also be other reasons why the DC-offset effect could come in handy; Observe the following small tutorial

The stereo-expander widens the samples being played in the track. This effect has only a proper value when working with stereo samples, or you use another DSP/VST effect that has a stereo effect-ouput. You always place the stereo expander behind the last effect you want to incorporate into the expansion. You can set the surround level to shift the dept of the audiofeed around or replace the signal in depth of the virtual dimension you desire to place the signal.

A gate is used to force a trackoutput to stay above a certain amplitude. If the signal fells below the threshold, the whole track-output will be cutoff.
This can be used to either remove a constant low noisefloor from a signal, or to cut of parts of complex signals (like drum loops) which fall below the threshold (gapping). Via the filters of the gates input, you can “analyze” only certain frequency ranges.
Use a gate to splice out (or duck) specific parts of complex drum loops, or to remove hiss and noise in silent parts of badly recorded material.
- Threshold - The dB value where the gate starts to let signal “thru”, it lowers the gaps.
- Attack - Determines how fast the Gate recognizes incoming signal peaks
- Hold - Determines how long the signal will be hold after the signal exceeded the threshold. Hold kicks in after the input signal moved above the threshold (attack is already applied). So its actually just delaying the release phase - holding the input above the threshold once it crossed the threshold.
- Release - Determines how low the Gate softens the incoming signal
- Floor - Volume in dB the output gets, when falling below the threshold
- Env. Input - Click to listen what gets feed into the Gate (to hear what the High Pass and Low Pass filters do)
- High Pass - Filter that will be applied to the analyzed (not outputted) signal
- Low Pass - Filter that will be applied to the analyzed (not outputted) signal
- Gate/Duck - the Duck mode reverses the above described threshold behaviour: The floor volume will be applied to signals falling below (not above) the threshold (it lowers the peaks).
Filters
Filters are pretty much around to amplify or attenuate various frequency rates. You can use the filter-effects to create specific effects (e.g. an old radio) or to remove irritating noise frequencies from bad quality samples. To explain the details of filters is really a topic any veteran sound-engineer can debate weeks about. I’ll give you a few hints whenever i think it is nessesary to fill them in. Either because the funtion is hard to understand or the tip is handy to burry in mind.

The filter-2 panel offers you a wide range of filterprocesses you can apply to your current track. Currently you see 5 low-pass filters in a row in the screenshot, but by clicking the selectors in the upper-right corner of the panel, you can browse through the following filter sets:
- Low Pass - LP 2x2 pole, LP 2 pole, LP biquad, LP Moog, LP Single
- High Pass - HP 2x2 pole, HP 2 pole, HP Moog, Band Reject, Band Pass
- Comp Mod - Distortion, Dist Low, Dist Mid, Dist High,A Mod
- EQ - EQ-15db, EQ-6db, EQ+6db, EQ+15db, EQ Peaking
- Cutoff - Sets the cutoff frequency for the current selected filter
- Reso / Q - Sets feedback resonance. note that using this slider in combination with other parameter effects may cause irritating sound-effects. So keep your volume low when fiddling with this slider.
- Inertia - Sets the speed of the cutoff effect slide from one frequency to the other. This is hard to explain without example, download and listen to both examples to hear the difference.
- Hihat track with LP 2x2 pole filter applied and automated cutoff freq. with inertia set to 0%
- Hihat track with LP 2x2 pole filter applied and automated cutoff freq. with inertia set to 100%

The 5-bands Equalizer.
Amplifies or attenuates the band of choice given in the panel.
The frequencies (left side of the slider) are adjustable, the peakwidth of the band is adjustable as well (right side of the slider)
The down arrow allows you to pick a different graphic scale:

The 10-bands Equalizer.
Amplifies or attenuates the band of choice given in the panel. If you need more precision than the 5 bands can offer, this panel should be able to do the trick.
The frequencies (left side of the slider) are adjustable, the peakwidth of the band is adjustable as well (right side of the slider)
The down arrow allows you to pick a different graphic scale:


The Mixer Equalizer is based upon a mixture of band-ranges.
The lo, mid and hi gain represent the various bandlevels that can be cutoff. The Mid frequency is the resonance factor, in combination with the quality factor. The quality factor affects the bass characteristics.
The down arrow allows you to pick a different graphic scale:


A dynamics processor that limits or prevents signals from peaking above a specified amplitude level.
A limiter works by compressing all sounds that peak above a specified threshold point. The average levels of your sound files can be raised several dB beyond the regular normalization level by using a limiter to compress the highest volume peaks. Using simple normalization alone to maximize the dynamic range has limitations. Because the normalization process simply calculates the available headroom, you are limited by the proximity of the loudest peak to the maximum cut-off point.
- Threshold - The dB amount of the maximum sound-output to hold (this value shall not be exceeded)
- Ratio - Threshold factor appliance, if threshold is exceeded, compression ratio is being applied to the exceeding peak.
To let the device limit the stream, pull the Ratio to it’s full value, the ratio will then display LIMIT in the value box. - Attack - The attack setting determines how long it will take the compressor to reach its full ratio of compression after it passes the threshold point.
- Release - Release sets how long it takes for the compressor to come back to its normal, uncompressed state.
- Post-gain - dB Offset base from where threshold will start to limit according to set percentage.
For more information about Compression / EQ try John Vestman’s page

A distorted or overdriven sound occurs when an audio signal is pushed above nominal limits for the amplifier or device it is driving. A form of distortion is clipping. More subtle distortion can occur in some amplifiers and devices and has become an art form in itself. Distortion and feedback are sometimes what is required.
The four buttons give you different distortion effects, they either supply a fuzz or a punch to the audio feed. Just pick a mode and use the sliders to drive the audio feed to your needs.
General parameters
- Wet out - Raise or lower the drive signal volume.
- Dry out - Raise or lower the bald sample volume.

Realtime bitcruncher, to make your samples sound like the old 8-bit sampler days….
- Bit cruncher - truncates each xth bit to match sample bitsize to the bit-width of the value
- Quality - Integrity of the sample wave (balances out noise-level).
- Noise - Amount of noise to add, lower quality level determines a higher noise integrity.
- Wet mix - Raise or lower the drive signal volume.
- Dry mix - Raise or lower the bald sample volume.

The delay set’s an echo of the sample, the amount of mseconds determines the delay between the sample played at initial time related to the echo that follows.
Note when setting the delay in rows, the amount of milliseconds will be calculated upon the current speed and bpm.
If you change those values, you need to apply a recalculation to get them synchronised to the pattern-speed again.
- L Delay - Delay of the left channel in msecs. Use the figure on the right to set the delay in rows (will be translated in msecs).
- R Delay - Delay of the right channel in msecs. Use the figure on the right to set the delay in rows (will be translated in msecs).
- L Feedback - Left channel sustain factor before echo fades out
- R Feedback - Right channel sustain factor before echo fades out
- Send - Volume of the delay effect
- Line Sync - Sync delay amount relative to the amount of pattern lines.
- L/R output pan - Set the stereoposition of the delay-signal
- Mute Source - Mute the dry audio-signal

A phaser shifts audio signals parallel to it’s channel, this produces the sound as being send through a tube or pipe.
A phaser comes from one direction.
- Ceiling - Bounce rate of the ceiling.
- Floor - Bounce rate of the floor
- LFO Rate - Frequency of the wave.
- Depth - Tube-size
- Feedback - Amount to modulate across it’s own signal
- Phase - Angle of shift
- Stages - Amount of phase waves being applied, the higher the more the amount. The lower the calmer the effect.

A flanger circulates audio signals parallel across it’s spectrum, This effect is great to produce e.g. sounds of engine rooms or plane engines with.
The flanger circulates between various directions.
- Amount - Modulation frequency.
- Rate - Circulation factor.
- Amplitude - Modulation amplitude.
- Feedback - Modulation feedback (from inbound to outbound).
- Delay - Phasing delay between modulation and original signal..
- Phase - Angle of shift.
- Filter type selection - Select type of filter to apply upon effect signal.
- Filter Freq. - Frequency range to apply filter to.
- Filter Reso. - Resonance amount generated within the effect signal.

Reverberation of the sound against roomborders.
- Wet Mix - Raise or lower the effect signal volume.
- Room size - Size of the room / effect-sustain
- Width - Width of the stereo signal. 0 = mono, 100 = full stereo
- Damp - Cutoff higher frequency bounces (sound-absorption level of the walls).
- Dry mix - Raise or lower the bald sample volume.

Reverberation of the sound against roomborders in a larger distance. Imagine your in a room that absorbs all sounds but the room attached to this one does not or a hall where the sounds in your direct surrounding are not bounced but further away they are reverberated, this is a bit the idea of playing an instrument on-stage in a concert hall, or someone in the room next to you is playing an instrument of which the walls do not absorb the sound completely.
You can achieve all these situations with this reverb effect.
- Duration - Reverberation time in msecs.
- Predelay - delay-time in msecs before the actual reverb is being bounced (effect is being played)
- Low cut - Low cut-off frequency of effect signal
- Low Gain - Boost low frequencies of effect signal
- Color - A combination between High cut and damp value of effect signal.
- Pan - Signal position.
- Width - Environment width of the effect signal.
- Wet mix - Raise or lower the effect signal volume.
- Dry mix - Raise or lower the bald sample volume.
Related topics
DSP-chain in general
Controlling meta/dsp/vst devices using Automation
Controlling meta/dsp/vst devices using pattern effect-commands
