Supplement to REAPER User Guide

The REAPER Cockos Effects

Summary Guide

version 3.04

December 2022

This document is intended to provide a summary of the various COCKOS plug-ins supplied with REAPER,
their broad purpose, the meaning of their various parameter controls and how to operate them. It serves to
supplement the User Guide, not replace it. It is not intended to serve as a comprehensive course on the
subject of audio effects and their many potential applications! For that reason, for commonly used types of
effects – such as equalizers and compressors – a basic understanding of the purpose of such effects is
assumed. More detailed explanations are given for those effects which are more REAPER specific, such as
ReaVocode and ReaVoice.

While every reasonable attempt has been taken to ensure accuracy in the preparation of this guide, the
author accepts no responsibility for any errors or the consequences thereof.

© Geoffrey Francis, December 2013, 2016, 2021

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 Table of Contents
1. The REAPER FX Plugin Interface.................................................................................7
2. ReaComp........................................................................................................................... 9
3. Sidechaining....................................................................................................................11
4. ReaControlMIDI............................................................................................................12
5. ReaDelay.........................................................................................................................14
6. Introducing Plug-in Pin Connectors.............................................................................16
7. ReaEQ............................................................................................................................. 17
8. ReaFir.............................................................................................................................. 19
8.1 EQ Mode...................................................................................................................................20
8.2 Gate Mode................................................................................................................................21
8.3 Compressor Mode....................................................................................................................21
8.4 Convolve L/R Mode.................................................................................................................22
8.5 Subtract Mode..........................................................................................................................22
9. ReaGate........................................................................................................................... 24
10. ReaInsert.......................................................................................................................26
11. ReaPitch......................................................................................................................... 27
12. ReaTune......................................................................................................................... 29
12.1 Tuning Mode...........................................................................................................................29
12.2 Automatic Correction Mode.................................................................................................30
12.3 Manual Correction Mode......................................................................................................31
13. ReaVerbate.................................................................................................................... 32
14. Channel Splitting: the JS Channel Mapper-Downmixer...........................................33
14.1 A Simple Example..................................................................................................................33
14.2 Parallel Processing.................................................................................................................33
15. ReaVerb......................................................................................................................... 35
15.1 Overall Controls.....................................................................................................................35
15.2 Echo Generator......................................................................................................................36
15.3 File...........................................................................................................................................37
15.4 Reverb Generator..................................................................................................................38
15.5 Filter........................................................................................................................................39
15.6 Normalize................................................................................................................................39
15.7 Reverse....................................................................................................................................40
15.8 Time/Gain/Stretch..................................................................................................................40
15.9 Channel Tools.........................................................................................................................41
16. ReaVocode..................................................................................................................... 42
17. ReaVoice........................................................................................................................ 43
18. ReaXcomp..................................................................................................................... 44
19. ReaLimit........................................................................................................................ 46
20. The JS Loudness Meter................................................................................................47

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21. General Dynamics........................................................................................................49
22. The JS Gain Reduction Scope......................................................................................52
23. The JS WigWare Multi-Channel Peak VU Meter......................................................53
24. ReaSamplomatic5000...................................................................................................54
25. ReaSynth....................................................................................................................... 56
26. ReaNINJAM.................................................................................................................57
27. ReaStream.....................................................................................................................57
28. The Frequency Spectrum Analyzer.............................................................................58
28.1 The Non-Nerds Guide to Frequency Analysis.....................................................................59
29. The Oscilloscope Meter................................................................................................60
30. Spectral Hold................................................................................................................61
31. Spectrograph Meter.....................................................................................................62
32. Super8 MIDI Controlled Synchronizer......................................................................63

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1. The REAPER FX Plugin Interface
All plug-ins in REAPER – including third party AU, VST and DX plug-ins – are displayed wrapped in REAPER's plug-
in interface. The various elements of this interface are explained within the User Guide, but for your convenience
a summary of these features is shown here.

Element Descriptions
Preset drop down list
Displays a list of presets supplied with the plug-in, and/or added by the user.
+ Menu button
Displays a menu for managing presets and (in the case of virtual
instruments) patches and banks. Can be used to store current FX
parameters as the default settings for that plug-in.
This includes options (shown right) for importing/exporting from/to
preset libraries and patch/bank files
Also gives access to Compatibility Settings menu.
Finally, you can choose to embed the GUI for this plug-in in the TCP
and/or MCP.
Wet/dry mix control
Controls how much of the signal without the plug-in applied (“dry”) is
mixed with a signal with the plug-in applied (“wet”).
For example, at 0%, only the dry signal is heard: this creates the same
sound assetting the plug-in to bypass. At 100% (the default setting) only
the wet signal is heard. At 50%, the wet and dry signals are mixed
together in equal proportions.
One use for this control is to moderate an effect that sounds overdone,
as can happen, for example, with delay, reverb or compression.
Alt click on this rotary to toggle delta solo on/off. Delta solo enables
you to hear only the difference between the wet and the dry signals.
Plug-in pin connector
This is a huge topic!
You can click on this button to open the pin connector matrix, which is examined later in this document and
explained in detail in the main REAPER User Guide.

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The pin connectors are mostly used when a track consists of more than 2 channels. They can be used to split an
audio signal between several channels, or indeed to rejoin them. They are also used when sidechaining (see
ReaComp).
Additionally, both MIDI and audio connections can be made from the context menu that is displayed by right-
clicking on this button. Audio can be received from, and output to, whichever channels (audio) or MIDI buses
(MIDI) that you specify.
This topic, including channel splitting and sidechaining, is covered in
the REAPER User Guide.
Your choice of stereo, multi-mono or multi-stereo will determine how
the audio is processed.
In stereo mode all master audio channels will be compressed based
on analysis of the loudness of all such channels summed together. In
multi-mono mode, each channel is compressed separately based on
an analysis of that channel. In multi-stereo mode, you will find that
each individual pair of channels channels will be compressed
independently based on an analysis of that stereo pair.
Pin to top toggle
Toggles keeping this FX window on top of other windows, even when
not selected. By default, this is set to off.
FX bypass toggle
Toggles the bypass state of the plug-in. When ticked, the plug-in is
engaged: when not ticked it is set to bypass. Default setting is
engaged.
UI toggle
Toggles display of any plug-in between its own GUI and a “vanilla” GUI.
Parameter menu
This is most commonly used to create for the last touched FX parameter a track control, an automation envelope,
parameter modulation or an assigned control surface control (such as a rotary or button). It can also be used to
assign your own preferred name (“alias”) to any parameter.
You can also (via the FX parameter list menu) assign a track control, automation envelope, parameter modulation,
control device control or alias to any other of the plug-in's parameters, whether it has been touched or not.
All of these actions are explained in the User Guide.
Miscellaneous
When a plug-in is displayed within an FX chain you can double click on its name to float the FX window. Right-
clicking on the floated title bar window gives yo options to unfloat the window, show the embedded UI in the MCP
and/or TCP, and to Send all keyboard input to plug-in.

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2. ReaComp
ReaComp is a compressor. Its most common use is for smoothing out variations in volume between the louder
and quieter parts of a track or folder. It can also be used to raise (or, less commonly, lower) the overall volume of
the compressed output of a track or folder.
The various parameters together define and determine the characteristics of the compression, such as at what
level(volume) it begins to be applied, how gently or harshly it is applied, and how suddenly or gradually it is
released.

Parameter Descriptions
Threshold
Determines the volume at which compression should be applied. In the example shown above, no compression
will be used when the volume of the track is below -21.4 dB.
Pre-comp
Allows the compression to gradually begin a specified number of milliseconds before the threshold is reached.
Attack Determines how quickly the compressor responds when the threshold level is reached or exceeded. A zero
setting means the full compression will be applied immediately and suddenly: the greater the number of
milliseconds specified, the more gradual the response.
Release
Determines how quickly the compressor reduces its gain reduction when the input volume decreases.
Auto release
This can facilitate using a very short release time without distortion.
Ratio
Determines the extent to which the compression will reduce the track's volume above the threshold. At 1:1 there
is no reduction. At 2:1, for every two decibels by which the dry signal exceeds the threshold level, the wet signal
will be increased by only one decibel. At infinity:1, the volume of the wet signal will be limited at the threshold
-level.
Knee size
The knee size determines the range of volume (rather than a specified amount of time) which the compressor will
use in applying the compression ratio. For example, with a ratio of 4:1 and a knee setting of 0 dB, the full 4:1
ratio will be applied as soon as the threshold is exceeded. With a knee size of 10 dB, the ratio will be gradually
increased from 1:1 at the threshold level, eventually reaching the full 4:1 when the volume exceeds the threshold
by a full 10 dB.

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Detector input
Determines whether the audio signal used to control the
compression will be the track's own signal (main) or an
auxiliary input signal from another track (enabled in the
pin connector settings (see right). Use of the plug-in pin
connector will be covered elsewhere in this guide.
For normal compression detector input should be set to
main. Use the auxiliary input setting for audio ducking –
that is, when you want the volume on one track to be
compressed according to the volume of a different track.
Audio ducking, which uses sidechaining, is explained in
the next section, and in more detail in the REAPER User
Guide.
Low pass filter
Specifies the highest frequency used to control the
compressor (frequencies above this are ignored for the detection). For example, a low pass setting
of 6kHz on a vocal track would ensure that the compressor is not triggered on a loud sibilance.
High pass filter
Specifies the lowest frequency used to control the compressor (frequencies below this are ignored
for the detection). For example, a high pass setting of 80Hz on a vocal track would ensure that the
compressor is not triggered on a loud plosive.
RMS Size
RMS (root mean square) is calculated according to a formula based on the square roots of a moving series of
values. Think of it as being conceptually similar to a moving average. If RMS is set above 0 ms, the higher the
setting the more extreme peaks and troughs in the audio signal will be smoothed out in determining the
compression that is applied.
Wet
This fader determines the level of the wet (compressed) audio stream.
Dry
This can be used to mix any required level of the dry (uncompressed) signal back into the compressed signal.
Preview filter
This can be used to audition the audio signal to assess the impact of your low pass and high pass filter settings.
Auto make up
When enabled, the level of the output volume will be increased to compensate for any reduction caused by the
compression.
Limit output
Prevents the output level from the compressor from exceeding 0.0 dB even after anygain is added by the Wet and
Dry gain faders. Different groups of parameter settings are more suitable for different types of instrument, and
even different styles of music. For example, for a more dramatic, percussive effect (such as you might use on a
rock drum kit) you would be likely to use more aggressive settings, such as a short attack and release times and a
harder knee. For a vocal ballad, however, you would be more likely to allow longer attack and release settings and
a softer knee.

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3. Sidechaining
Sidechaining allows you to use the audio signal of one track to control the behavior of a plug-in such as a
compressor or noise gate on a different track. An example might be when you want an instrument mix
compressed by a lead vocal.
In that case you would first create a send from channels 1 and 2 on the vocal track to channels 3 and 4 on the
instrument submix or folder.
First, you would create a send from channels 1/2 of the Vocal track to channels3/4 of the instrument folder.
You can do this in the track routing window of the Vocals track or in the track routing matrix. Both these options
are shown below.
Adding a send using the track routing window

Adding a send using the Routing Matrix

You can now set the detector input ReaComp or ReaGate to Auxiliary Inputs and you’re ready to go. In this
example, if using ReaComp, the threshold setting will apply to the volume of the vocal track, while the
compression settings will be applied to the instruments mix.

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4. ReaControlMIDI

ReaControlMIDI is a plug in for use with MIDI items or tracks which contain MIDI items. It can be used to create
and send MIDI control messages in real time.

Parameter Descriptions
MIDI Channel
Select All MIDI channels or any individual channel. Separate instances of ReaControlMIDI can be applied to
different channels.
All Notes Off
Sends All Notes Off message – useful for dealing with any “hanging” MIDI device.
Show/Hide Log
Opens and closes display of MIDI activity log: in the example shown above, the log is displayed.
Clear
When the log is visible, the Clear button will clear its contents.
Enable Bank/Program
A toggle which allows a bank and program to be selected and used.
Bank
Drop down list to select MIDI bank.
Program
Drop down list to select a program from the current bank.

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MSB/LSB
Displays MSB (most significant byte) and LSB (least significant byte) values for currently selected bank/program.
Load Bank
Opens Explorer/Finder to allow you to select a bank to be loaded.
Transpose
Option to transpose notes by specified number of semitones.
Snap to scale
Toggles set to scale option on/off.
Scale value
Select required scale (e.g. C, G).
Scale type
Select as required:
Scale type (e.g. Major, Pentatonic)
or Chords (e.g. Major Triad, Minor Triad)
or Mode (e.g. Dorian).
Scales can also be loaded from a.ReaScale file.
Enable Control Change
A toggle which allows up to five control change parameters to be controlled from the ReaControlMIDI interface. In
the example shown above, three of these are used, Volume, Pan and Pitch Wheel.
Enable Raw Mode
A toggle to allow raw mode (0 to 127) access to all CC messages.
Edit Sysex
Opens a window for entering system exclusive messages.
Re-send
Resends sysex messages.
Enable Always send sysex on playback start
Toggles this option on and off.
Enable Log all notes-off messages Toggles this option on and off.
Like all other COCKOS plug-ins, any of the ReaControlMIDI parameters can be used to create automation
envelopes. Any envelopes that you do create in this way can be set to Write mode to record parameter fader
movements and/or value changes in real time as you play the track back, and then to reproduce these
movements when it is played back again.

An example of this is shown above, where two automation envelopes have been written to a MIDI track, one for
each of two ReaControlMIDI parameters. The first of these produces several volume changes, the second
produces changes to the pan setting. These changes are also displayed as events in the ReaControlMIDI window.

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5. ReaDelay

ReaDelay, as its name implies, is used to add delay to a track or media item. Its possible uses are many and vary
from adding a few milliseconds of simple delay to a thin vocal (to fatten it up a little), to creating an echo effect to
add to a timpani or other low frequency percussion instrument (to create an atmosphere of impending tension) to
creating a bouncing ping pong effect. This is one effect that more than most really opens up avenues for you to
be creative.
REAPER's ReaDelay gives you multi-tap options. This means that you can create several discrete pages of delay
settings independently of each other apply them all at the same time to to a track or media item. The default
screen shows only one tap (labeled “1” above) but you are able to add more if you wish.

Parameter Descriptions
Enabled
Toggles enabling/disabling the currently selected delay tap.
Solo active tap
Toggles soloing the currently selective tap (settings page). When any one tap is soloed, all other taps will be
ignored.
Length (time) / Length (musical)
Use one of these settings to specify the length of delay required, either in time (milliseconds) or fractions of a
note. It would be unusual to use both of these faders on a single delay tap.
The human ear will not normally distinguish a delay setting of less than 7 ms as a discrete sound. Such a setting
can therefore be helpful in making a track sound “fatter”.
Feedback
Feeding the delay signal back into the original dry signal can help blend the two signals together, thereby possibly
making the wet signal less “echoey” and perhaps creating some interesting effects.
But beware! Too much feedback can have unpleasant consequences, both in terms of sound quality and your
ears!
Lowpass filter
Specifies a frequency above which the audio signal passes thru without the delay effect being applied. For
example, a low pass setting of 5,000 Hz would ensure that the delay was added only to frequencies below 5,000
Hz.
Highpass filter
Specifies a frequency below which the audio signal passes thru without the delay being added. For example, a
high pass setting of 10,000 Hz would ensure that the delay was added only to frequencies above 10,000 Hz .

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Resolution
This fader can be used to reduce the bit resolution of individual taps. In the example shown above, the default
setting of 24 bit matches the resolution of the recorded material. Lowering the delay tap bit resolution can
emulate the sound of early digital delay effects boxes.
Stereo Width
When working with stereo material on a track, bus or folder, the stereo width fader can be used to narrow the
stereo image of the delayed signal.
Volume
Adjusts the level of volume for the individual delay tap.
Pan (unlabeled)
Located immediately to the right of the volume fader, this adjusts the pan setting for this individual delay tap to
the left or right.
Add tap
Adds another page of delay settings.
Delete tap
Removes the current page of delay settings (but you must leave at least one page).
Reset all
Resets the FX to its default settings, including restoring the same number of taps as
in the default settings.
Wet
Determines the output level of the mixed wet signal of all delay taps.
Dry
Determines the output level of the original (undelayed) signal, mixed in with the wet
signal.

The next section will show you how to use REAPER’s pin connectors to add ReaDelay to a track and then send the
wet signal (i.e. the delayed signal) to a different track. This technique can be applied to many plug-ins, including
(but by no means limited to) delays, reverbs and choruses . ReaDelay is used purely for the purpose of
illustration.

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6. Introducing Plug-in Pin Connectors
Sending the output of an FX can open up interesting mixing possibilities: for example, you might wish to pan the
dry and wet signals independently of each other. You can do this using REAPER’s pin connectors.
Clicking on the 2 in 2 out pin connector button will open the window
shown here (right). This indicates that your audio signal is currently
coming into the plug-in on channels 1 (left) and 2 (right) and going
out on the same channels.
The track channels dropdown can be used to add more channels.
Moreover, by clicking in the individual cells of the matrix you can
redirect output to any (and as many) channels as you wish.

Thus, as shown below, you can increase the number of channels to

four (from the drop down) and redirect output of the delayed signal
from channels 1 and 2 to channels 3 and 4.

All that remains for us to do is to open the routing window for our destination track (in this example, track 6) and
add a receive from the original vocal track. The signal is received from channels 3/4 and output on channels 1/2.
If you wish, you could for example now add more FX to (in this case) track 6 such as compression and/or EQ in
order to shape the delay effect but not the original dry signal on (in this case) track 1.

Later in this guide, we’ll be looking at other models using REAPER’s pin connectors.

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7. ReaEQ

ReaEQ is a parametric equalizer which by default contains four bands. However, you can add more bands or
remove any existing bands if you wish to do so. It is used to make adjustments to the frequency spectrum of a
track or media item. By cutting or boosting any required frequency or range of frequencies you are able to change
some of the characteristics of an audio signal. This can be used to compensate for existing deficiencies (such as
too much bass or top end, or to remove noise, hum or rumbling), or to enhance a track's sound (such as by
adding more presence or warmth).
As with many effects, use ReaEQ gently, especially at first. Applying equalization too aggressively can make a
track or media item sound worse rather than better.
Parameter Descriptions
Numbered tabs
In above example, there are four bands, numbered 1 to 4. Click on any numbered tab to select that EQ band.
Enabled
Toggles the currently selected band between enabled and bypass state.
Type (drop down list):
Determines the type of band used for the selected EQ band. Main options are:
Low shelf: Applies gain or attenuation only to those frequencies below the selected frequency.
High shelf. Applies gain or attenuation only to those frequencies above the selected frequency.
Band. Applies gain or attenuation across a frequency range and centered on a selected frequency. The width
or narrowness of the range is determined by the bandwidth setting.
Low pass. Allows frequencies below the specified frequency to pass thru, but tapering down to eventually
eliminate frequencies above that specified.
High pass. Allows frequencies above the specified frequency to pass thru, but tapering down to eventually
eliminate frequencies below that specified.
All pass. Allows all frequencies to pass, but changes the phase relationship between the various frequencies.
Notch. Allows thru all frequencies except a range centered on the selected frequency. The width of the range
is determined by the bandwidth setting.
Band Pass. Cuts everything above and below a range centered on the selected frequency. The width of the
range is determined by the bandwidth setting.
Parallel Band Pass. Parallel band passes are run separately and then summed: the remaining filters run in
series after that.

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Log-scale automated frequencies
Toggles this option on and off. When enabled, it produces a more useable scale for frequency automation
envelopes, so that the lower frequencies are allocated a larger area of the automation envelope area. This is
achieved by using a logarithmic scale rather than a linear scale for the frequency envelope.
Frequency (Hz)
Sets the frequency for this band that is to be attenuated or to which gain is to be added. Its value is displayed to
the right of the fader both in Hz and as a MIDI note value. This can also be set by clicking and dragging the point
on the graph.
Gain (dB)
Determines the amount gain or attenuation (expressed in decibels) that is to be applied to this band. This can also
be set by clicking and dragging the point on the graph.
Bandwidth (oct)
Determines how large (wide) or small (narrow) will be the frequency range to which the attenuation or gain will
be applied. Measured in octaves.
Add band
Adds another band (page) to the ReaEQ settings.
Remove band
Deletes the currently selected band (settings) page.
Reset defaults
Restores this ReaEQ instance to its default settings.
Show tabs
Toggles between display of the ReaEQ tabs and graph only.
Show grid
Toggles display of the graph grid on and off.
Show phase
Toggles this option on and off. When enabled, the graph will illustrate
the extent of phase-shift caused by your EQ settings.
Context menu
Right-click over any of the numbered points on the graph to display a
context menu of band settings (shown right).
Most commands duplicate functions already explained, such as
changing band type, or adding a band. Of special interest is the option
Flip all bands, which changes attenuated frequencies to gain, and vice
versa.
Gain
The vertical Gain fader on the right raises or lowers the volume of the
overall wet output audio signal.

Note: REAPER's Action List includes the assignable action Track: Insert
ReaEQ (track EQ) which can be used to insert an instance of ReaEQ in
any track.

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8. ReaFir

ReaFir is a multi purpose plug featuring five different modes – EQ, Gate, Compressor, Convolve L/R and Subtract.
Each of these functions is entirely independent of the other. If you wish to use more than one of its functions (for
example, compressor and EQ), then you need to use a separate instance of ReaFir for each function.
Each mode has its own specific controls: in addition there are some parameters that are common to the different
modes. The information that follows has been adapted from the REAPER Wiki.

Shared Controls - Parameter Descriptions

Graph display window
This window shows the analysis of the audio being processed, as well as the user defined parameter curve. This
window by default shows frequencies from left to right (representing 0 Hz to 24kHz), and volumes from top to
bottom (representing by default +24dB to -90dB).
The red line represents the user defined parameter curve, the dark yellow represents the input signal, and the
bright yellow represents the output signal.
Volume range
The two edit boxes to the right of the graph display window (top and bottom) can be used to change the default
volume range. By default these are set at +24 and -90dB.
Reset button
Resets the user-defined parameter curve to a flat line at 0dB.
Edit mode
This drop-down list draw very complex curves.
Points (smooth) - the default: in this mode you can create any number of points that define the curve, the
lines between the points are automatically smoothed and curved.
In points modes:
Add points by clicking on the red line on the graph.
To delete a point, hold Alt and click on it.
Double click on a point to return its gain value to 0 dB.
Right click on a point and choose Set to set its gain and/or frequency.
Hold Shift while dragging a point up/down to adjust gain without changing frequency.
Hold Shift while dragging a segment of the curve to move that segment up or down.
Hold Ctrl while dragging on the curve to move whole curve up or down.

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FFT Size
This controls what sized blocks are used for processing. In general larger FFT (Fast Fourier Transform) sizes allow
more precise effects, and use more CPU. Some modes, however, may be affected in other ways by the FFT size.
For example, in the Compressor and Gate modes, larger FFT sizes result in a larger effective attack/release time.
Quality dropdown
This can be used to lower ReaFir's performance quality to make it compatible with earlier versions. Unless you
really need to do this, leave the setting at its default which is Best quality.
Mode
Selects ReaFir operating mode – EQ, Gate, Compressor, Convolve L/R or Subtract.
Show analysis floor
Disable this if you do not care to see the approximate spectrum of the signal displayed on the graph.
Analysis floor edit box.
Lets you choose the lowest value of analysis shown. In general you should keep this set to the same as the low
range of the curve rangeadjustment (above), but sometimes you might want it to be different. For example,you
might want the adjustment range to be 12..-12, but still display the analysis down to -90dB).
ReaFir's analysis display is an approximation: this is a consequence of prioritizing processing efficiency over
display quality.
Output Gain
Lets you adjust the volume of the output, after all processing has occurred.
Reduce artifacts
By default ReaFir operates in a non-linear FFT-based filtering fashion. The downside is that in certain instances
the filter can introduce aliasing artifacts, in exchange for extremely high filtering effectiveness.
If you check this box, ReaFir will operate as a finite impulse response (FIR) filter, meaning it will be limited in its
effectiveness by the length of the filter, but will respond linearly and without artifacts.

8.1 EQ Mode
In EQ mode, the red adjustment curve applies gain or attenuation at different frequencies.

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8.2 Gate Mode

In Gate mode, the Gate floor fader sets a benchmark threshold, so that when the audio signal falls below that
threshold the sound is filtered out. In the example shown here, a benchmark threshold is set at -40.0 dB, to
eliminate unwanted background sounds like (perhaps) breathing. This threshold applies to the level of the track or
media item as a whole, not to individual frequencies.
In addition, the red adjustment curve can be used to gate specific frequencies or ranges of frequency. In the
example above, this is used to gate out any possible low end rumble and very high end sizzle. The faint dark
yellow line shown on the graph at around -50 dB at around 0 Hz to 75 Hz illustrates this.

8.3 Compressor Mode

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In Compressor mode, ReaFir acts as a multi band compressor. However, there is a notable difference between a
regular multi band compressor and ReaFir. With ReaFir, you can choose more specifically which frequencies you
want to compress without having to worry about crossover. In addition, ReaFir in compressor mode can have
hundreds or even thousands of bands which can all be compressed separately.
The red adjustment curve represents the threshold so that when the signal goes above it, it is compressed by the
ratio specified by the Comp Ratio fader.
Note that the FFT size set affects both the size of the bands used for compression, as well as the effective
attack/release times (a larger FFT size has longer attack/release times).
The ratio can be set with the ratio fader, and can anywhere in the range of a gentle 1.01:1, a hard 100:1, or less
than 1:1, giving per-band upwards expansion capability.
Because of the flexibility of specifying threshold per-band, you are able to choose exactly which frequency band
gets compression, and can do some very interesting things with it.
Keep in mind that if you lower the threshold too much, you will hear audible distortion from the fastattack/release.
You can define a threshold only below 100hz to compress bass frequencies.
The FFT window will show you how much gain reduction is happening in bright yellow - the original signal's
frequencies will still show as dark yellow.
Compressing higher frequencies alone can be an effective de-esser. You may wish to do this only at the points
where the sibilants occur in the recording.

8.4 Convolve L/R Mode

In Convolve L/R mode, ReaFir filters the left channel's signal with the right channel's signal, and applies the
adjustment curve as an EQ control.
In Convolve L/R mode you will hear the "like" frequencies from the Left and right channels together, and any
dissimilar frequencies will be faint or non-existent. This is useful to check the overall balance of a mix to see what
is happening sonically - especially if there is too much of a frequency band (in other words, an extreme build-up
of like frequencies).
This mode has no special controls of its own. However, you can make EQ adjustments using either precise lines or
points. This can help you make decisions about what needs to be done within a mix to balance it out.

8.5 Subtract Mode

ReaFir's subtraction mode is excellent for Noise Reduction! It is especially useful when there is a noise (such as
perhaps a faint hum or hiss) that is present in the background of a recording of a track.

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When you select Subtract mode, you will notice a new button that says Automatically build noise profile. To
use this function, enable it while auditioning a section which contains only the noise. You will see the profile being
built (see example above). Once you are happy with the noise profile, un-check Automatically build noise profile,
stop playback and save the preset.
Alternatively, you can create the subtraction curve yourself, by playing the noise, switching to "precise" mode, and
drawing on the graph right above the noise.
You can then do a number of interesting things with the noise profile. For example, hold Ctrl, click and drag the
noise profile to raise or lower the entire profile (the higher you raise it, the more artifacts you will hear).

Still in Subtract mode, play the track (see example above). The profiled noise will be removed from the processed
signal.

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9. ReaGate

ReaGate is a noise gate whose most common use is to filter out sound when the volume falls below a specified
threshold level. This can help to eliminate unwanted sounds, such as perhaps a vocalist drawing breath between
verses.
One feature of ReaGate is that like ReaComp it gives you the option of using an auxiliary signal, so that the gate
on one track can be controlled by an audio signal on another track.

Parameter Descriptions
Threshold
This is the unlabeled vertical fader on the left. It is used to specify the input signal level (volume) below which the
gate should be engaged.
Pre-open
This is the first of five parameters that together shape exactly how the noise gate behaves in opening and closing.
Pre-open offers you the option to look ahead a specified number of milliseconds so that the Attack will be
activated ahead of the volume actually falling below the Threshold.
Attack
This determines the speed with which a closed gate reopens when the volume rises above the threshold. The
lower this setting the more quickly it will go from letting thru 0% to 100% of the signal. Too fast a setting here
can sometimes result in audible clicks.
Hold
This determines how long the gate is held open (before closing) when the signal falls below the Threshold.
Variations in Hold time can be used with Reverb plug-ins to help create gated reverb.
Release
This determines the speed with which a gate closes when a signal falls below the threshold level (after the Hold
time).
Hysteresis
Think of this as fulfilling a similar function to the thermostat on a central heating system. It specifies a margin of
flexibility around the Threshold level, to prevent the gate from constantly opening, closing, and “chattering.”
Detector input
Determines whether the gate is controlled by the track or media item's own audio signal (Main Input) or by the
signal from another source (Auxiliary Input).

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Lowpass
Specifies the highest frequency used to engage the gate (frequencies above this are ignored
for the detection).
Highpass
Specifies the lowest frequency used to engage the compressor (frequencies below this are
ignored for the detection).
RMS Size
Specifies the period of time to be used in calculating RMS (Root Mean Squared) values. In general, the higher this
value, the less “fussy” will be the gate.
Send MIDI on open/close
Enable this toggle to send MIDI note on/off messages when the gate opens or closes.
MIDI note and channel
Note number and channel to use if Send MIDI option (above) is enabled.
Wet
Sets the level of the processed signal in the output.
Dry
Sets the level of the pre-gate unprocessed signal in the output.
Noise
Allows a level of noise to be mixed back into the output.
Preview filter output
When detector input is set to Auxiliary, enabling this toggle lets you to the output of the sidechain source.
Invert gate (duck)
Enabling this toggle causes the polarity of the wet signal to be reversed relative to the dry signal. Whenever the
gate is open, the two signals will phase-cancel. This enables ReaGate to be used in conjunction with a sidechain
signal for 'ducking.”

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10. ReaInsert

ReaInsert lets you use an external hardware effects device (such as a reverb unit) with your recorded audio
material. It can also be used to send MIDI data to an external synthesizer for processing. The device needs to be
connected to your sound card's inputs and outputs. Its controls are for the most part straightforward.

Parameter Descriptions
Send vol
Controls the level of the signal being sent to the hardware effect.
Hardware Sends audio:
Left / Right
Select the hardware outputs (from your sound card) to be used for sending left and right channels respectively.
Hardware Sends MIDI:
MIDI Device
Channels From/To
Select the installed MIDI device (if any) to which you are sending data. Specify both source and destination
channels.
Hardware Returns:
Left/Right
Specify the hardware inputs (on your sound card) to be used for receiving the audio signal back from the
hardware effects device or synthesizer.
Automatic device latency adjustment
If enabled, automatically adjusts playback to compensate for any latency caused by the device.
Automatic Delay Compensation
The Ping detect button can be used to estimate the degree of delay required to compensate for any latency
introduced by the external device.
Return vol
Controls the level of the signal returning to REAPER from your hardware device.

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11. ReaPitch

ReaPitch is used to raise or lower the pitch of an original sound. It can be used, for example, to create harmony
effects on vocals, or to fatten the sound of an instrument. As well as being able to make changes to pitch, you
can also adjust the formants of your material. Formants are perhaps a more difficult concept to grasp than pitch:
you can think of them as being the spectral peaks of a sound, that is, the elements that give a sound such as an
individual voice the special characteristics by which it is uniquely identified.
Like ReaDelay, the default settings for this plug-in include a single page of settings (shifter), but you can add
more shifters if you wish, each with its own settings. The volume strength and pan position of each shifter can be
adjusted independently of each other.
It’s worth noting that ReaPitch is another plug-in that lends itself well to being used with pin connectors to
direct the wet signal to a separate track (see section 6.). This would potentially give you more options (for
example, panning) for mixing vocal harmonies. You could even have multiple instances of ReaPitch, each directed
via separate paired channels (3/4, 5/6, 7/8, etc.) to its own track.

Parameter Descriptions
Wet (volume)
The combined level of processed (wet) sound (from all shifters) that is added to the output.
Dry (volume)
The amount of unprocessed (dry) sound that is added to the output.
Enabled
A toggle to enable/disable individual pages of shifter settings.
Solo active shifter
A toggle to solo the currently selected shifter (settings page), effectively disabling all others.
The next seven controls listed are used to raise or lower pitch or formants. These controls are cumulative. For
example, setting Shift (cents) to +50 and Shift (semitones) to +1 would raise pitch by 1.5 semitones.
Shift (full range)
Raise or lower pitch by the specified number of semitones.

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Shift (cents)
Raise or lower pitch by the specified number of cents. A cent is 1/100th of a semitone.
Shift (semitones)
Raise or lower pitch by the specified number of semitones.
Shift (octaves)
Raise or lower pitch by the specified number of octaves.
Formant shift (full)
Raise or lower formants by the specified number of semitones.
Formant shift (cents)
Raise or lower formants by the specified number of cents. A cent is 1/100th of a semitone.
Formant shift (semitones)
Raise or lower formants by the specified number of semitones.
Volume
Determines the level of volume from this shifter (settings page) that will be added to the processed (wet) signal.
Pan (unlabelled)
This fader (to the right of the volume fader) sets the pan position of the output of this shifter (settings page) in
the processed (wet) signal).
Add shifter
Adds another settings page.
Delete shifter
Deletes current settings page. You must leave at least one page.
Reset all
Restores default settings, including the default number of shifters. The remaining parameters (below) are set for
the ReaPitch instance as a whoe, not for individual shifters.
Autocorrect master playrate change
Enable this toggle to ensure that your ReaPitch settings are applied regardless of any changes made to the master
play rate.
Pitch Shift Mode Algorithm
A drop down list of choices. Project Default will use the algorithm selected in the Project Settings dialog. Other
options at time of writing are:
Sound Touch
Simple windowed
Elastique 2.2.8 Pro
Elastique 2.2.8 Efficient
Elastique 2.2.8 OLO
Elastique 3.3.3 Pro
Elastique 3.3.3 Efficient
Elastique 23.3.3 SOLO
Rubber Band Library
Stereo
A toggle to enable/disable stereo output

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12. ReaTune
ReaTune has three main modes – tuning mode, automatic pitch correction, and manual pitch correction. The
required mode is selected by clicking on of the three tabs located near the top of its interface.
To tune an instrument such as an acoustic guitar, connect the guitar via a lead to your sound card and set up a
track ready for recording in the usual way. Insert ReaTune into the track's input FX chain. Enable input monitoring
and make sure the track is armed for recording.
To use ReaTune in one of the other two modes, insert it into the FX chain of track or media item, then select the
required mode.

12.1 Tuning Mode

Parameter Descriptions
Window size
Determines the number of samples at a time on which analysis is performed. The larger the window size, the
lower the frequencies that can be analyzed but the more CPU will be used.
Overlap
Determines whether the windows are overlapped, and if so by how much. For example, on a 50ms window, at 1x
20 blocks per second will be analyzed, at 2x 40 blocks per second, and so on. This increases CPU usage but will
provide faster tuning times.
Reference
The reference note. By default this is the A above middle C.
Note display window
This shows the note that ReaTune is processing. The note name is displayed in the upper window, and the pitch
of the input in the lower window: the position of the note in this window indicates how in-tune is the input note.
Send MIDI event when pitch changes
When enabled, this toggle causes a MIDI event message to be sent whenever the pitch changes. This, for
example, can be used to send the MIDI event data from an audio track prepared for recording audio (such as a
guitar) to another track with a VSTi and some other virtual instrument such as a cello or viola. You could then use
your guitar to play the virtual instrument in real time!

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12.2 Automatic Correction Mode

Parameter Descriptions
The Automatic Pitch Correction option should be ticked to enable this function.
Key name
From the drop down list select the key, in the range A thru G#.
Key mode
Select a mode/scale from the drop down list. Available modes are Major, Minor,Chromatic, Dorian, Phrygian,
Lydian, Mixolydian, Locrian, Arabian and Egyptian. The default setting is Chromatic.
Notes allowed
All/any of the 12 notes shown (from A thru G#/Ab) can be enabled or disabled for pitch-correction purposes. For
example, if you were to enable only, say, C, then all notes would be “corrected” to C. This would most likely
produce an excellent example of singing out of pitch as is humanly possible! By default, all notes are enabled.
Attack time
Determines the time between the start of the note and when ReaTune commences any required pitch correction.
Stereo correction
A toggle to enable or disable stereo mode.
Algorithm
Select the algorithm best suited to the signal being corrected. For example, for vocals this is likely to be elastique
2.1 soloist. Options are Project Default (as specified in the Project Settings dialog), Sound touch, Simple
windowed, elastique pro, elastique efficient and elastique SOLO, 2.2.8 or 3.3.3, or Rubber band library.
Parameter
The list of available parameters will depend on which algorithm is selected. The parameter is used fine tune the
way the selected algorithm is applied.
Clear history
The graphical display uses vertical bars to illustrate the effect of the pitch correction(see above). The Clear history
button clears the existing graphical display from the window.

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12.3 Manual Correction Mode

In manual correction mode, you are able to correct pitch errors by using horizontal bars (see above).
To draw a pitch correction bar: click and drag the bar from left to right or right to left with your mouse.
To move a bar up or down: click and drag up or down.
To extend or shorten a bar: click and drag the start or finishing point (small circle) left or right.
To delete a bar: select it with the mouse and press Delete.

Parameter Descriptions
Manual correction
This toggle enables/disables manual correction mode. After making your corrections, this can be used as a means
of comparing the signal before and after your changes.
Track pitch
This toggle allows the graph to be updated as REAPER plays.
Update
Instantly updates/refreshes the graphical display.
Clear
Clears the graphical display.
Prevent octave shifts
Unexpected harmonics in an audio signal can sometimes cause ReaTune to interpret the note as being shifted up
or down an octave. Enabling this toggle prevents these notes from being shifted.
Prevent overlapping segments
When enabled, this toggle prevents you from drawing pitch correction lines that overlap each other.

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13. ReaVerbate

ReaVerbate is the simpler of the two Cockos reverb pluh ins, the other being ReaVerb, which we will get to later.
ReaVerbate can be used to help create a feel that your song has been recorded in an environment such as a hall
with natural reverberation rather than in a cold studio. Don’t be deceived by its apparent simplicity. If you
experiment, you will be surprised at its capabilities.
Its essential characteristics are described below.

Parameter Descriptions
Wet
Determines the amount of the processed (wet) signal mixed into the outgoing audio stream.
Dry
Determines the amount of the unprocessed (dry) signal mixed into the outgoing audio stream.
Room size
Specifies the size of the room whose natural reverberation you are aiming to imitate.
Dampening
Moderates the reverb effect to allow for the fact of items such as curtains and carpet that might be present in the
room.
Stereo Width
This control can be used to reduce the stereo field of the reverb effect.
Initial Delay
Allows for a delay ODF a specified number of milliseconds before the reverb is produced. Higher settings can
sometimes create a feeling of more space.
Lowpass
Used to ensure that the reverb is not applied to higher frequencies (above the lowpass setting).
Highpass
Used to ensure that the reverb is not applied to lower frequencies (below the highpass setting). Reverb will be
applied to all frequencies in the range between the highpass and lowpass filter settings.

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14. Channel Splitting: the JS Channel Mapper-Downmixer
Using the Pin Connector with the Channel Mapper

14.1 A Simple Example

In this section we will be taking another look at REAPER’s FX plug-in pin connector by examining an example of
how you can split FX across different channels to achieve some interesting results. For the sake of simplicity we’ll
work with one FX -ReaVerbate. We’ll be laying the reverb on fairly heavily at first, purely as a means of helping
you to understand what’s going on. If you can listen thru headphones when doing this you will notice the effect
more obviously.
Take a vocal track – a copy of one that you have used for something else maybe.
Insert into this track’s FX chain an instance of ReaVerbate.
Adjust the ReaVerbate settings so that the effect is quite powerful – in
particular, set a large room size of about 80 and an initial delay of about 60.
As you listen, the reverb effect should be more than obvious.
Now comes the clever bit! Open the ReaVerbate pin connector. Adjust the
settings so that an audio stream is sent from this plug-in to channels 3/4, as
illustrated here (right).
Now play the track. You won’t hear any reverb! That’s because we have sent
it off on channels 3/4 – we now need to be able to mix it back in again
So … we can control the amount of reverb on the track with a pan control.
The main trouble with that, of course, is that you might want to do this
without having to send the signal fully left or right.
Solution? Use a channel mapper/mixer!
Add the JS: Channel Mapper-Downmixer to
the track FX chain. This is shown here
(right). In this examples the options to Pass
through, and also User mix are both
selected. These options are explained in the
summary at the end of this section.
The grid at the bottom needs to be set as
shown, so that both the dry signal on
channels 1/2 and the reverb signal on
channels 3/4 are both passed to channels
1/2 on the track.
By adjusting the four vertical faders as you
prefer, you can ensure exactly the required
amount of reverb is added not only to the
track overall, but also individually to each of
the left and right speakers.
And, as a bonus … those volume faders can
be automated and/or have controls added to the TCP. Right-clcik on the Param button and follow the menu.
This is a relatively simple example that uses only one FX. Later you will see how this technique can be applied to
an FX chain with any number of FX.

14.2 Parallel Processing

In this example we will be building on the above example so as to be able to run two (or indeed more) FX chains
in parallel. Here’s an example of what this means.
You might want to apply both EQ and compression to a track. But which should come first? One possible answer
is … both and neither! By splitting the chain into two streams you can apply each effect separately, joining them
together at the end.
Let’s work thru an example. This might, for example, be a vocal or guitar track, but it could be anything.

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Add one of the FX to the track’s FX chain – let’s use ReaComp for this example. Because ReaComp allows for
sidechaining, it might be wise to leave channels 3/4 and use channels 5/6 instead.
Make what adjustments you need to the compressor’s
parameters (e.g. threshold, ratio, knee size). Click on the 4
in 2 out button to display the plug-in pin connector.
Set the number of channels to 6 and direct output to 5/6 as
shown. (For this example, you can ignore the designated
auxilliary inputs).
Now add ReaEQ to the FX chain and make adjustments as
required. Leave the pin connector inputs and outputs at
their default settings – 1/2 and 1/2 .
After ReaEQ, add the JS Channel Mapper-Downmixer to the
FX chain.
Select the options to pass through and for the user mix.
Lower the faders for channels 3/4 all the way down – you
want to mix together only the EQ’d signal on 1/2 and the
compressed signal on 5/6.

Shown here are the channel mapper-

downmixer settings for this example,
but before any adjustments have been
made to the vertical faders. Audio
streams from channels 1/2 and 5/6 are
both output through channels 1/2.
Your FX chains can be as simple or as
complex as you like.
For example, suppose that you wanted
to add some delay on 5/6 to the
compressed signal. You could simply
add ReaDelay to the chain immediately
after ReaComp, making sure you set its
pin connector inputs and outputs both
to channels 5/6.
Be prepared to experiment and be
adventurous! You’ll be surprised at what
you can achieve.

Parameter Descriptions
Unmapped outs
Zero outs can give you greater control control over your outputs. Any inputs that are not assigned outputs in the
matrix will not be passed thru to the track. This makes you think about where you are going to direct outputs in
the matrix, and ensure that the matrix unambiguously and accurately describes your routing.
Pass through means that any unmapped input channel will automatically be passed through to the same output
channels. In the second example above, as long as pass through is enabled you need not have specified an output
for channels 1/2 – they would have been sent out via 1/2 anyway. However, had you not specified an output for
5/6 then that output would not have been heard as REAPER would attempt to output the audio streams to
channels 5/6. Th
Downmix
You need to set this to User mix if you are to have control over the channel volume faders.

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15. ReaVerb
ReaVerb is an extraordinarily powerful reverb plug-in. You start with a blank canvas and build up your reverb
effect using a combination of modules, listed in alphabetical order:
channel tools, echo generator, convolution reverb file, high/low
pass filters, normalization, reverb generator, reverse reverb,
time, time/gain/stretch.
Modules are added by clicking the Add button and making a selection from the
drop down list. Where you have more than one module, the order can be changed
by dragging up or down your displayed list of selected modules. A module can be
removed by selecting it and clicking on the Remove button.
You can use as many modules in your reverb chain as you wish, but don't fall into
the trap of thinking that because all these options are available, the more modules
you use the better will be the results! Quite possibly a single module, such as
reverb generator or a convolution reverb file, will be quite sufficient to meet your
needs. The method used to “chain” different reverb modules together is
explained, with examples, in the main REAPER User Guide. This chapter details the
different parameters and controls of the various modules.
Shortly, we'll take a look at each module individually, but first let's examine the overall ReaVerb environment,
starting with the blank canvas.

15.1 Overall Controls

Parameter Descriptions
Wet
Determines the level of the total processed (wet) signal included in the output.
Dry
Determines the level of the original unprocessed (dry) signal included in the output.
Add
Adds a module to ReaVerb – Channel Tool, Echo Generator, File, Filter, Normalize, Reverb Generator, Reverse, or
Time/Gain/Stretch. Modules are listed in the Impulse generation window. Drag and drop to change their order.
Remove
Removes whichever module is currently selected in the Impulse generation window.

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Width
Adjusts (restricts) the width of the reverb stereo field.
Pan
Pans the overall reverb signal more to the left or right.
Pre-reverb
Any setting above 0 milliseconds causes the reverb effect to be added to the audio signal ahead of the dry signal,
by the specified number of milliseconds.
Performance Options:
Max FFT FFT (Fast Fourier Transform) filters are a complex mathematical topic. Changing the FFT size
changes CPU usage. A lower FFT setting means a higher CPU usage, but read more evenly. Higher FFT sizes
will consume less CPU but result in more latency. FFT size will therefore affect performance and possible
dropouts, but it should not affect the sound itself. If unsure about changing it, accept the default setting.
ZL
Use this option to enable zero latency: useful when tracking to monitor reverb levels.
LL
Use this option to use an extra thread to improve low latency performance.
Set all
This applies performance settings to all instances of ReaVerb in the current project.

15.2 Echo Generator

The display in the window above the parameter controls gives excellent visual feedback as to how the various
parameter controls affect the echoed signal. In addition, the following comments should prove helpful.
This example shows the Echo Generator being used on its own. You can also obtain interesting results in
combination with other modules – for example, placing it after a Reverb Generator.
Parameter Descriptions
Start position
How much time (in milliseconds) is to pass after the sound before the echo commences.

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Length
How long the overall echo effect is to last.
Spacing
The amount of time that passes between individual echo sounds within the total echo length. The smaller this
setting, the more dense will be the echo.Error Introduces a degree of variation between individual spacing.
Errist
This modifies the effect of the error setting by introducing variations.
Start volume
The volume (strength) of the echo signal when it starts.
End volume
The volume (strength) of the echo signal when it finishes.

15.3 File

Parameter Description
Browse
Opens Explorer/Finder to enable you to select a different reverb impulse file.
Mix to mono
A toggle to select stereo or mono output. Default is stereo.
Normalize/attenuate impulse dropdown
Presents a menu of options for applying normalization and attenuation to the reverb impulse. Normalization will
apply a constant amount of gain to the reverb impulse, and will have the effect of increasing its average
amplitude. Attenuation (e.g. to -18 dB) will will reduce the audible level of the reverb impulse. This can prevent
the original signal from being “drowned” in too strong a level of reverb. If unsure of what to do, leave this at its
default setting.
Destination channel offset
This ca be used to direct the reverb output to channels other than the ReaVerb input channels.

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Generate test tone …
Creates a test tone wave file from the convolution impulse reverb file.
Deconvolve
Enables you to use a test tone file to deconvolve (that is, to reverse the effects of convolution) an existing file.

15.4 Reverb Generator

The display in the window above the parameter controls gives excellent visual feedback as to how they variously \
affect the echoed signal. In addition, the following comments should prove helpful.

Parameter Description
Start position
This can be used to introduce a delay (in milliseconds) between the incoming audio signal and the time the reverb
commences.
Length
This defines the time it takes for the reverberation to decay (or fade away). The higher the decay setting, the
more likely you are to need to increase the Dampening setting in order to prevent the sound from being too
muddy. Let your ears be the judge!
Volume
The volume of the reverb signal.
Room Size
Specifies a smaller or larger room whose environment you are aiming to simulate. In general, a larger room size
will produce a more noticeable reverb effect.
Dampening
This control simulates the extent to which a room's natural reverberation is affected by being absorbed by factors
in the room as walls, carpets and curtains.
Stereo Width
This can be used to modify the stereo image of the reverb output.

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15.5 Filter

This module is used to add a lowpass

filter and/or a highpass filter to a
reverb signal. These allow frequencies
above that specified by a low pass filter
and below those specified by a high
pass filter to pass thru the plug-in
without the reverb effect being applied.
Use the faders, or type directly into the
edit boxes, to specify the required
frequency range.
Put more simply, the reverb will be
applied only to that range of
frequencies between the low and high
pass filter settings. For example, setting
a lowpass filter of 100 and a highpass
filter of 5000 would have the effect of
applying the reverb only to frequencies
between 100 and 5000.
This can be used, for example, to
prevent reverb from being applied to
very low (bass) frequencies.
Note that there would be little point in
attempting to use a Filter module by
itself: it works in conjunction with other
modules, such as File or Reverb
Generator to modify their effect.

15.6 Normalize

This module can be placed in ReaVerb's

file chain to normalize the signal that is
passed to it be whichever modules
occur before it (in the example above,
Reverb Generator and Filter).
A constant amount of gain can be
added to the signal to bring the peak
amplitude (volume) up to whatever
level is specified by setting the fader.

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15.7 Reverse
This module can be used to add to
the audio signal a level of reversed
reverb, determined by the level of the
Wet and Dry faders.
The Dry fader determines the level
used of the signal that is being fed
into this module. Notice that this is
not “dry” in the most commonly used
sense of the word, as it will include
any and all effects added by any
preceding modules – in the example
shown here, file and a filter.
The Wet fader controls the amount
(volume) of reversed reverb that will
be added to the signal.

15.8 Time/Gain/Stretch
This module can be used in conjunction
with other modules such as File or
Reverb Generator.
For example, if you have an impulse file
whose sound you like but perhaps want
to vary the length of the reverb tail, you
can achieve this by using
Time/Stretch/Gain to lengthen or
shorten the effect.

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Parameter Descriptions
Start position Ensures that the time stretch/gain is not implemented until a specified number of milliseconds into
the reverb impulse.
Max length Use this control to change the maximum length of the impulse response.
Gain Specifies in decibels the required output level for this module.
Stretch Use this to stretch or shrink the incoming reverb impulse. A value below 1.00 will shrink, above 1.00 will
stretch. Be cautious here and make any adjustments, at least at first, in relatively small increments.
Pre-silence Allows for a period of silence (in milliseconds) before the time stretch/gain is implemented.
Width Allows you to moderate the stereo width of the outgoing signal effect.
Pan Allows you to adjust the position of the module's effect more to the left or right.

15.9 Channel Tools

The vast majority of users can ignore this feature which would only be needed in a limited number of advanced
and complex applications. However, for the record here is a summary of its controls.

Specify your source channel, channel count and destination channel.

Then, from the drop down list, select the Action that you require.
Swap swaps the two channel ranges.
Copy: copies the channels from the source to the destination.
AddTo: sums the source to do the destination (if you took two reverbs and ran them in parallel, this
would be the resulting value).
SubtractFrom: similar to AddTo, but flips the sign of the first reverb impulse response.
Convolve: combines the two impulse responses in series (e.g. running the output of reverb into
another), this results in the length of the impulse response being the summed length (two one second
impulses convolved would produce a two second impulse)

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16. ReaVocode

A Vocoder can be used to add a robotic, synthetic, sci-fi tone to a recorded track. You can set it up so that any
track is modified by another. One common application uses a synthesizer to modify the sound of a vocal track.
First, you should take the time to understand some terms that are used:
The detector track is the track which is being modulated – in this example, this would be the vocal.
The modulator track (also known as the carrier) is the track that is used to modify the sound of the
detector track. In the example that follows this would be a synthesizer.
Here's an example of how you could set this up.
 Record the vocal on a track by itself.
 On a different track, use a VSTi to record the tune. Use a fairly bland sound – you can change this later.
 Make sure that both of these tracks have four channels. and that the their master/parent sends are both
disabled: to do this, click on a track's I/O button and display its routing window.
 Create a new track and insert ReaVocode into its FX chain. This is the Vocoder track.
 Create a send from the detector track (vocal) to Channels 1/2 of the Vocoder track.
 Create a send from the modulator track (synth) to Channels 3/4 of the Vocoder track.
 Play the song, adjusting the vertical faders on the Vocoder to get the required balance.
As you can see, there are few controls on this plug-in. They are explained below.

Parameter Descriptions
Wet
Determines the amount of the processed (wet) detector signal mixed into the outgoing audio stream.
Dry
Determines the amount of the unprocessed (dry) detector signal mixed into the outgoing audio stream.
Mod dry
Determines the amount of the unprocessed (dry) modulator signal mixed into the outgoing audio stream.
Bands
Specifies the number of frequency bands to be used in modulating the detector signal.
Invert signal inputs
A toggle to swap the roles of the two incoming signals: the modulator becomes the detector and vice versa.
Stereo
A toggle to enable/disable stereo output.
Once you've got the hang of it, you can explore the creative possibilities at your leisure. For example, use a vocal
track to modulate a guitar to create a singing guitar effect.

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17. ReaVoice
ReaVoice is a MIDI controlled multi-voice pitch shifter. You can use it in conjunction with a recorded vocal track to
create pitch shift harmonies. In overview, this is how it works:
 Record the Vocal Track.
 Insert a new track immediately below the Vocal Track.
 Create a send from the Vocal Track to the new track. Initially at least, this should be Pre FX.
 Insert ReaVoice into the FX Window of the new track.
 Arm this track for recording. Select your MIDI keyboard as the Input Device and turn Input Monitoring
on.
 Mute all tracks except these two.
 Play the song. As you do so, play the keyboard, experimenting until you find an appropriate range of
notes.
 Work out what you want to play, press W to return to the beginning, then Ctrl R to record. Stop
recording when finished.
 If you do not have a MIDI keyboard, you can either use REAPER’s Virtual keyboard, or you can enter
the notes by hand using the MIDI Editor.
 If you wish, you can record more than one take, selecting Play All Takes for the MIDI track items.

Parameter Descriptions
Wet
Determines the amount of the processed (wet) signal that will be mixed into the outgoing audio stream.
Dry
Determines the amount of the unprocessed (dry) signal that will be mixed into the outgoing audio stream.
Gain
The amount by which the volume of the incoming signal is increased or decreased.
Max voices
The number of vocal harmonies required, maximum is eight.
Attack
Determines how quickly the plug-in responds to the incoming audio signal.
Sustain
Determines for how long the harmonies will be held.
Release
Determines how quickly effects generated by the plug-in will take to die down after the sustain time is finished.

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MIDI channel
Selects the channel(s) used for the MIDI material (0 = all).
Center note
Selects center note (in semitones).
Pitch shift Algorithm
Selects the pitch-shift algorithm to be used – Sound Touch, Dirac, Simple windowed, elastique Pro, elastique
Efficient, elastique SOLO, Rubber Band Library or Project Default. Project default is specified in project settings.
Pitch shift Parameter
Selects parameter to fine-tune the behavior of the selected algorithm. If Project Default is selected, this option is
not available, but will be determined by the Project Settings.
Stereo
A toggle to enable/disable stereo output.

18. ReaXcomp
ReaXcomp is a multiband compressor. Broadly speaking, its controls are similar to those found on ReaComp,
except that you are able to split the frequency spectrum into several bands and compress each band individually.
The example below shows a compressor with four bands.
Another feature of ReaComp is that you can make adjustments to some of its parameters by using your mouse on
the plug-in's graph. These options are explained below.

Parameter Descriptions
Numbered tabs (shown as 1 to 4 above)
Selects a band and displays that band's settings. In the example shown above, band 2 is selected.
Active A toggle to enable or disable the current band.
Top Frequency Sets the highest frequency of this band. The lowest frequency is determined by the top frequency
of the previous band. In the case of band 1, the bottom frequency is 0. A band's top frequency can also be
adjusted by dragging left or right the thin vertical blue line that separates bands on the graph.
Threshold Sets the volume level at which this band of the compressor will be engaged. Note that this is the
threshold for the band, not for the audio signal as a whole.
A band's threshold can also be adjusted directly on the graph, by dragging its band number up or down.

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Ratio
This determines the amount of compression applied to the band. For example, at 2:1, for every 2 decibels by
which the volume of the band on the dry signal exceeds the threshold, the processed signal will be increased by
only 1 decibel. Setting a ratio below zero will make the band act as an expander rather than a compressor.
Auto make up gain A toggle which when enabled will automatically increase the gain on this band when its
volume falls below the threshold.
Program dependentrelease
A toggle which when enabled ensures that attack and release times are modifiedaccording to the frequency and
amplitude of the audio signal.
Feedback detector
A toggle which when enabled will prevent feedback.
Knee
The knee size determines the range of volume (rather than a specified amount of time) which the compressor
band will use in applying the compression ratio. For example, with a ratio of 4:1 and a knee setting of 0 dB, the
full 4:1 ratio will be applied as soon as the threshold is exceeded. With a knee size of 10 dB, the ratio will be
gradually increased from 1:1 at the threshold level, eventually reaching the full 4:1 when the volume exceeds the
threshold by a full 10 dB.
Attack
Determines how quickly the compressor band responds when the threshold level is reached or exceeded. A zero
setting means the full compression will be applied immediately and suddenly: the greater the number of
milliseconds specified, the more gradual the response.
Release
Determines how quickly the compressor band responds when the volume drops below the threshold level. A zero
setting means that the compression will be fully and instantly released. A higher setting ensures that the release
will be more gradual.
RMS Size
RMS (root mean square) is calculated according to a formula based on the square roots of a moving series of
values. Think of it as being conceptually similar to a moving average. If RMS is set above 0 ms, the higher the
setting the more extreme peaks and troughs in the audio signal will be smoothed out in determining the
compression that is applied.
Log-scale automated frequencies
Toggles this option on and off. When enabled, it produces a more useable scale for frequency automation
envelopes, so that the lower frequencies are allocated a larger area of the automation envelope area. This is
achieved by using a logarithmic scale rather than a linear scale for the frequency envelope.
Add band
Adds a new band (tab).
Delete band
Deletes the current band (tab). You must leave at least one band.
Reset to defaults
Restores this instance of ReaXcomp to the plug-in's default settings.
Solo current band
Disables all bands except the current one.
Tabs
Toggles on and off display of tabs and their parameter controls.
Grid
Toggles on and off display of grid lines on the graph.
Gain
Raises or lowers the volume of the plug-ins overall output (not individual bands).

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19. ReaLimit

The purpose of a limiter is to enable you to lift the overall volume of your mix while at the same time ensuring
that it does not exceed a defined threshold. One special feature of ReaLimit is that it gives you in real time visual
feedback of how your output is being affected by the limiter.
Parameter Descriptions
Threshold
This determines the level at which you want the limiting to begin: the lower the threshold setting, the greater will
be the gain.
Brickwall Ceiling
This defines a maximum level above which the signal will not be raised.
Link
If enabled, this links the threshold and ceiling faders, so that both will be moved together in the same direction if
you move either one.
True Peak
This causes true peak values to be used for the display: for an explanation of “true peak” see the section The JS
Loudness Meter.
Release
This determines the rate at which limiting is released after the signal drops back below the threshold. The greater
the release time, the smoother and more gradual is the transmission.
Performance
This can be set to high quality or low latency. In most cases you can leave this at its default setting.

Note:
The red lines in the display represent the boundaries within which the signal is being limited. The blue graph
above the waveform indicate where the signal has been limited and by how much. The areas on the waveform
shaded white indicate the extent to which the signal has been limited on those occasions.

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20. The JS Loudness Meter

Note:

The display of parameter controls is

turned on and off by clicking on the ?
In the lower right hand corner.
If the entire window (controls and
graphic display) is not visible you may
need to resize the window, by clicking
and dragging up from the top border or
down from the bottom border.
The JS Loudness Meter displays visual
feedback about LUFS (loudness unit full
scale), RMS (root mean squared) and
True Peak loudness. Insert it into any
track or master just as you would any
other plug in.
Before examining its parameters and
controls in detail you will need to
understand some technical terms, and
in particular Peak and True Peak, RMS
(Root Mean Squared) and LUFS
(Loudness Units to Full Scale).

Peak: Highest actual level of the audio signal. Observing and controlling the peak is important to prevent
distortion or clipping. Peak levels will change frequently and rapidly during a song – for example, when a cymbal
is struck.
True peak: Highest level of the oversampled signal. Intended to estimate the highest level the signal could reach
after resampling to any sample rate. You would generally expect this to be higher than the peak.
Root Mean Squared: This is a continuous average voltage measurement for electrical signals, usually measured
at 300 ms. It offers a good indication of the average loudness of a track. As shown in the above illustration, RMS
values can be expected to be lower than Peak levels and to fluctuate less dramatically. As a general rule, the
greater the difference between Peaks and RMS the greater the dynamics.
Loudness Units referenced to Full Scale: The loudness of the digital signal, after applying filters intended to
adjust for how humans perceive different frequencies. You can think of it, if you wish, in terms of the Fletcher-
Munson equal reference contours: in brief, this takes into account that the human ear hear low and high
frequencies less well than it hears mid range frequencies.
These days LUFS is generally regarded as the preferred industry standard.
Now let’s examine these parameter controls …

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Parameter Controls
Peaks:
Whether to display true peaks, true peak clips, peaks or peak clips, or turn peak display off.
RMS momentary and/or integrated:
Momentary is measured over a short period, such as 400ms: integrated reflects the measurement of the entire
audio signal, from beginning to end.
LUFS momentary and/or short term and/or integrated:
Momentary is measured over a short period, such as 400ms: short term is measured over a 3 second window:
integrated reflects the measurement of the entire audio signal, from beginning to end.
LRA loudness range:
The dynamic range of the entire audio signal, measured as the statistical difference between the lowest and
highest LUFS-S measurements. Whether or not to display this.
LUFS alerts:
Whether or not to show LUFS alerts, and if so at what levels and which colors.
Reset on playback start:
Whether to reset existing meter settings when a new playback is commenced.
Force mono analysis:
Whether to analyze data in mono.
Text size and Y-axis scaling
Defines the font size to be used for text and the scale to be used for the vertical axis.

The small orange symbol above the ? can be used to toggle between stereo (default) and mono analysis.

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21. General Dynamics
General Dynamics is a highly configurable general purpose dynamics processor that can function as a compressor,
expander, and/or gate. The user can define a mapping of input loudness to gain directly by drawing lines and
curves with the mouse.
General Dynamics offers extensive controls on the input (sidechain, feedback-input, gain, RMS, attack/release), as
well as the output (gain attack/release, wet/dry). Instead of using faders to determine settings such as threshold
and ratio, you draw them on the graph with your mouse.

Above are shown two relatively simple examples of how this plug-in can be used. The first (left) uses first an
expander for the quieter passages, and a compressor for the louder passages.
The second applies sledgehammer compression at a specified volume.
In many scenarios, using General Dynamics will still allow a high amplitude signal thru. You might wish to consider
adding a limiter such as the JS Event Horizon Limiter to your FX chain, to prevent clipping.
These are just two simple examples. It's up to you how you use this! For example, you could try this: add it to the
master mix, draw a horizontal line at -12db across the whole UI, then turn the volume down of all of your tracks.

Basic mouse operations are:

Click and drag Draw straight line.
Ctrl click and drag Draw freehand.
Shift drag A blur tool. Changes sharp angle into a smooth transition (see first example above).
Click while drawing line Use Alt drag to change curve, Ctrl to commit.
Alt drag Erase
Shift right-click drag Zoom (drag up to zoom in, down to zoom out) – or use mousewheel.
Right-click drag Move graph.

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These next two examples might be considered a little less conventional.

The first could be used (depending on the material to which it is applied) to add clarity to a track or mix.
The second (and it is only an example, not a recommended model for any particular application or project) shows
how you can create some quite funky sounds. In this case we have zoomed in quite close to enable ourselves to
work more precisely in the area that interests us.
Detector Input
But wait, there's more! You can change the detector input setting from main to aux or feedback.
Main can be considered as “normal” and is the default setting. General dynamics' behavior is controlled by the
audio signal on the same track or item as the plug-in. This is how a compressor, expander or gate is most
commonly used.
Aux means that the plug-in's behavior is controlled and triggered not by the audio signal on the same track, but
by that routed in from a different track. For this to happen, a send needs to be created from the controlling track,
out from channels 1/2 and in thru channels 3/4 on the destination track. This process is often referred to as
“sidechaining”. You can read more about sidechaining (and other audio routing techniques) in Section 3, and in
the REAPER User Guide.
Feedback should only be used with extreme caution, as the act of feeding a signal back on to itself can produce
dangerously loud signals, both to your equipment and your hearing (especially if you are using headphones).
Feedback uses the output of the compressor as the detector input. Depending on the curve drawn and
attack/release/RMS settings, this can do different things - when expanding, GD will be more inclined to stay
applying gain for longer (since the output will remain higher as the input falls), or when compressing GD will be
more inclined to stop applying gain sooner (since the output will be lowered by the compression).
You should use caution with respect to the curves used in both Aux and Feedback modes, because the detector
input is so strongly decoupled from the signal being processed, it is very easy for large amounts of gain to be
applied to an already-loud signal.

Page 50
Fader Controls
Finally, let's take a look at the General Dynamics fader controls, which help to shape and fine tune the plug-
in's output. For the most part, these are standard features of most common compressors, expanders and/or
gates.
Parameter Descriptions
Detector Gain Can be used to increase or decrease the volume of the signal being fed into the General Dynamics
plug-in. This directly affects only the level of gain that the detector “hears”, not the level output.
Detector RMS size
RMS can be used to slow down the responsiveness to the signal. This levels off peaks and troughs in the incoming
signal. A larger RMS signal will lead to a smoother signal, meaning that ordinarily the plug-in will be engaged less
dramatically.
Input look ahead time
A higher look ahead time can be handy, for example, for dealing with sudden loud transients that go beyond the
plug-in's input attack settings. It causes the compressor to be engaged before the audio reaches this loud peak,
thus ensuring that the amplitude never exceeds a desired level.
Input attack The input attack setting determines how quickly the compressor is engaged. A short attack is usually
preferred for percussive and other sounds with sudden peaks. A longer attack is usually recommended, for
example, for a mix or for instruments which peak more gradually.
Input release The release setting determiners the rate at which the compression effect is released. This helps to
determine how the compressor shapes the sound. A short release time might be preferred for percussive
instruments, but too short a release time can sound unnatural, whereas a longer release time will often sound
smoother.
Gain attack This determines the rate at which gain is applied to the signal. See also note below.
Gain release This determines the rate at which signal gain is released. See also note below.
Wet mix This determines the level of the post-FX signal that will be output.
Dry mix This determines the level of the pre-FX signal that will be output.
Note: Gain attack and gain release roughly correlate with input attack/release in normal gate/compression/
expansion use. However if you ever have negative relationships (increase of input volume = decrease in
output volume - a line that goes from upper left to lower right on the graph), the gain A/R can be used to
control those curves differently (e.g. attack/release relationships are somewhat backwards).

Page 51
22. The JS Gain Reduction Scope

The JS Gain Reduction Scope helps you to analyze how your dynamics are being affected by the plug-ins in your
FX chain. The first two columns represent the dry, unprocessed signal. The last two represent the signal after it
has been processed thru your FX chain. In order to
be able to do this we first need to split our signal
into two chains.
First, set the number of track channels to four.
This is done in the track’s routing window,
remember!
Then, as the first item in the track’s FX chain,
add the Channel Mapper Downmixer, Make sure
that the output from channels 1/2 goes to both
channels 1/2 and ¾, as shown here.
Next, add to the chain whichever FX you
require, ensuring in each case that the signal
comes in fro channels 1/2 and goes out to
channels 1/2 only.
This will ensure that channels 1/2 will carry the wet
stream and channels 3/4 will carry the dry stream. It is important that you
do not mix these streams back together!
Now, at the end of your FX chain, insert the JS: Gain Reduction Scope. Click
on the 4-in button to display its plug-in pin connectors. Make sure that the
input from channels 1/2 is output to channels 1/2 only, and that the input
from channels 3/4 is output to channels 3/4 only, as shown here.
When you play the track, the two audio streams will each be analyzed
separately, and the differences will be displayed. Now we just need to
examine what exactly is being displayed!

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Parameter Descriptions
Parameter values are set by left clicking and dragging up or down over it. Any parameter can be reset to its
default value by double-clicking on it.
Meter range
The dB range to be used for the vertical display meters. This can be set anywhere between -2dB and – 150 dB.
Drag all the way down to turn meter display off altogether.
RMS
The unit of measurement to be used for calculating RMS. The valid range is from 0.02 ms up to 500 ms.
Range
The dB range to be used for the graphical display. Valid range is from -6.0 dB to -100.0 dB. Drag up/down on the
graphical display itself.
Length
The length of time displayed at any one time in the graphical display. Valid range is 0.13 ms to 3.0 seconds.
Hold
Click and hold for a second then drag down to the graph and release to “freeze” display. Drag up/down to change
to dB range shown in the frozen or left/right to go back or forward in time. Click again on “hold” to unfreeze.

23. The JS WigWare Multi-Channel Peak VU Meter

Although not a Cockos plug-in, it’s

worth taking a quick look at the JS
WigWare Multi-Chanel Peak VU meter,
as it gives you a simple view of a
comparison between up to 16 different
channels in a track. In the example
shown here, the track has six channels.
Its controls are simple and self-
explanatory.
Although simple, it can be quite useful if
you are working with multi-channel
tracks.

Page 53
24. ReaSamplomatic5000

ReaSamplomatic5000 is a plug-in that can be

used to play a sample under MIDI control. You
can use multiple instances for multisamples
patches and drum kits (as explained in the
REAPER User Guide).

Parameter Descriptions
Mode
A drop down list to select Freely configurable shifted, Sample (ignores MIDI note) or Note (semitone shifted).
Import item from arrange
If an audio item is selected in the arrange window, this is used as the sample.
Browse
Opens Explorer/Finder for you to select the sample file that will be played.
[list]
Toggles open a window where you can build, store and manage a list of items.
Volume
Sets the volume of the sample when triggered at the highest velocity.
Velocity maximum
Sets the maximum velocity level.
Pan
Used to pan the sound in the stereo mix towards left or right.
Min vol
Sets the volume of the sample when triggered at the lowest velocity.
Velocity minimum
Sets the minimum velocity level.
Note start
Defines the MIDI note for which note-on messages will trigger the sample.
Pitch@start
Specifies the pitch offset of the sample when it starts (Freely configurable and Note modes only)
Note end
Defines the MIDI note for which note-on messages will stop the sample.
Pitch@end
Specifies the pitch offset of the sample when it ends (Freely configurable and Note modes only)

Page 54
Pitch offset
Specifies a value (in semitones) to change the pitch of the sample.
Pitch bend
Determines the amount of pitch variation allowed.
Attack
Determines how quickly the volume of the sample rises when triggered.
Sustain
How loud the sound rings out while the note is held.
Decay
Determines how quickly the sound tails away after the release period.
Release
Determines how long the sound is held.
Note-off release override
A toggle which allows note-off release to be overridden by a specified time (ms).
MIDI Chan
The MIDI channel(s) on which the plug-in listens for note-on and note-off messages.
Portamento
Allows you to add a portamento effect.
Probability
Determines the probability factor for beats, so that every cycle can be made different.
Round-robin
A toggle which when enabled allows the same version of a sample to be used when it is repeated. It makes the
application of probability somewhat deterministic. If you have, for example, a 20% chance, round-robin will
ensure a response exactly every 5th note. With Round-robin disabled, each note is independently randomized.
Remove played notes from FX chain MIDI stream
A toggle option which gives you the option of not passing played notes thru the FX chain.
Obey note-offs
A toggle to allow note-off messages to be obeyed or ignored.
Loop
Allows a properly configured sample to play with its loop section repeating until a note-off message is received.
Xfade
Can be used to create a cross fade when looping.
Loop start offset
Can be used to create a time offset, specified in milliseconds, when looping.
Max voices
Determines the maximum number of samples that can be triggered concurrently without forcing a sample that is
already playing to stop. The parameter control knob can be used to select any number between 1 and 16, or any
other number up to 128 can be entered manually.
Cache samples smaller than
Specifies a sample size below which data is to be cached. Using cache can improve speed of performance: how
much cache is available to you will depend on your computer's RAM size.
Resample mode
A drop down list with a ranger of choices varying by quality from Low to Best or by speed from Fast to Extreme.
You also have the option of using the Project Default, as specified in your Project Settings dialog.

Page 55
25. ReaSynth

ReaSynth is a synthesizer. Inserted into a track's FX chain

itcan be played using a MIDI keyboard or, indeed, REAPER's
virtual keyboard. You can also insert empty MIDI items and
use the MIDI editor to create your music and MIDI
messages.

Its control parameters are described below.

Parameter Descriptions
Volume
Sets the volume output level. A sensible approach is to start with this at a relatively low settingof around -10dB or
so, then fade it up to suit.
Tuning
Adjusts note or pitch of the output.
Attack
This represents how long (in milliseconds) it takes for the sound to rise from an initial value of
zero to its maximum level.
Decay
This is the length of time takes for the sound to fall off to the sustain level.
Sustain
The volume level at which the sound is sustained before it is released.
Release
The time taken after the sustain period for the sound to fall away to zero.
Wave shape parameters
The synthesizer sound is made up from a mixture of various wave forms of
different shape. The remaining controls together shape this sound.
Square mix
How much square wave is added to the sound.
Pulse width
The width of pulses: the wider the pulses, the smaller will be the gaps
between them.
Sawtooth mix
How much sawtooth wave is added to the sound.
Triangle mix
How much triangle wave is added to the sound.
Extra sine mix
How much extra sine wave is added to the sound.
Extra sine tune
Any adjustment to be made to the pitch of the additional sine mix.

Page 56
26. ReaNINJAM

ReaNINJam is not an audio effect and therefore outside the scope of this document. It is software that can be
used in conjunction with NINJAM. NINJAM is an innovative software program that allows musicians to collaborate
in "fake time" over the Internet. The software connects to a central server where participants can share audio and
text information, and has the ability to record both the local and remote channels of each "session" (audio only).
REAPER has the ability to import these recorded session files to allow you to edit and mix at your leisure. "Fake
time" means that some players will hear a delayed version to which they play along.
ReaNINJAM is described in more detail in the main REAPER User Guide.
See also http://www.NinJam.com and http://www.cockos.com/wiki/

27. ReaStream
ReaStream is not an audio effect and therefore strictly
speaking outside the scope of this document.
It supports streaming from host to host of audio and or MIDI
over a LAN segment. The plug-in needs to be inserted and
enabled with a common identifier on both systems. The option
to send should be enabled on one machine, and to receive on
the other.

Page 57
28. The Frequency Spectrum Analyzer

Frequency analyzers are used to model the frequency spectrum of audio signals in such a way as to depict visibly
various elements of the signal such as distortion and harmonics. Frequency is plotted on the horizontal axis and
bandwidth on the vertical axis. The Frequency Spectrum Analyzer controls are:

Parameter Descriptions
Phase
A toggle: click to determine whether phase noise is displayed or not.
Windowing functions
Right-click for a menu with a choice of windowing functions. You have a choice of which windowing function you
want applied in performing FFT analysis. These are blackman-harris, blackman, rectangular or hamming. The
default is blackman-harris.
Fast Fourier Transform (FFT) size
Right click for a menu to select ant FFT size between 16 and 32768. We're entering a complex world of
mathematics here, and unfortunately there's no easy way to explain fourier transform to the uninitiated in a few
words! As a starting point, think of it as being a technique that analyzes a signal by deconstructing it into the
frequencies that make thatsignal up. One use for this is to make the analysis of that signal easier.
Changing the FFT size changes CPU usage: a larger FFT size provides more frequency resolution, however it also
reduces the time resolution, meaning that more samples are used for analysis. You get more of an average of a
longer period of time. A larger FFT also makes the graph inherently slower to react to changes in the input signal.
Integrate
7The default setting is 0 milliseconds (0 ms). Click and drag up to increase this, down to lower it again. At higher
settings the display will slow down and smooth out. Double-click to restore the default setting.
Slope
Click-drag here to set the slope within a range of 0 dB to 12 dB per ocave.
Floor
This setting determines the quietest extent of the graph. The default is -120 dB. Click-hold-drag down to increase
this range, or up to reduce it. Double-click to restore the default setting.
If you're technically minded and want to understand these concepts a bit better, read the notes on the next page.

Page 58
28.1 The Non-Nerds Guide to Frequency Analysis
Frequency analyzers show the frequencies and their levels in an audio signal. They display the signal not as level
against a time axis (time domain, like the REAPER peaks display), but as signal level against its contained
frequencies, or frequency spectrum (frequency domain). This is like splitting an audio signal into its frequency
components in the same way as splitting light into a rainbow with a prism.
Why do this?
Some aspects of audio which are hard to see in the time domain are very easy to see in a frequency-domain
display, such as harmonics.
How do they work?
Joseph Fourier's basic theorem states that any signal in the time domain can be represented by the
weightedsum of pure sine waves of all frequencies. This means any signal can be thought of, equal to, and
reconstructed rom, varying amounts of sine waves. The “weighted sum” bit could be drawn as a line graph of
levels for those sine waves at their frequencies, showing how much of each frequency the signal contained, and
that graph would actually be the signal's “spectral content”.
To display the spectral content, we use an algorithm called a “Fast Fourier Transform” (FFT) to quickly
approximate the levels (weightings) at a set of frequency points. The FFT is simply a clever set of operations
which implement Fourier’s basic theorem; it inputs a number of consecutive samples of the audio and analyses
them to given levels at frequency points. The more samples used, the bigger the FFT size is said to be, and the
finer the resolution of its frequency points, traded-off against time taken.
For fs=48kHz audio, No. freq-points is half FFT size, highest point at Nyquist freq.
1024 sample size yields 512 points from (fs/1024) 46.9Hz to 24kHz taking (1024/fs) 21.3ms
2048 sample size yields 1024 points from (fs/2048) 23.4Hz to 24kHz taking (2048/fs) 42.7ms
4096 sample size yields 2048 points from (fs/4096) 11.7Hz to 24kHz taking (4096/fs) 85.3ms
etc...

Fast Fourier Transform (FFT) size:

The number of consecutive samples used per analysis. A larger number gives a higher resolution of frequency
points and a lower first frequency point, but takes longer per analysis cycle (slower update rate).
Example: FFT size of 1024 would give 512 frequency points and a lower-frequency limit of around 40-50Hz with
an update rate of about 20ms (50Hz) for standard audio sample rates.

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29. The Oscilloscope Meter

An oscilloscope displays and analyzes the waveform of electronic signals. While the track is playing it displays a
graph which depicts instantaneous signal voltage across time. The visible waveform can then be subjected to
analysis, forexample, of aspects such as frequency, amplitude and time intervals. As with the spectrum analyzer,
frequency is plotted on the horizontal axis and bandwidth on the vertical axis.

Parameter Descriptions
Retrig
Triggering is the process of synchronizing the horizontal sweep of the oscilloscope with the proper point of the
signal being analyzed. This control can be set to determine whether a signal is triggered on ascending flank or
descending flank, or instantly, or by any. Click on this control to cycle between options.
Range
This determines the limits of the dB range displayed in the chart. The default setting is 0.0 dB but this can be
modified. Click-hold-drag over this control, upwards or downwards to change the range setting as required.
Double click to return to the default setting.
Length
This determines the length displayed in the chart. The default is 10 Hz. Click-hold-drag up or down to change this.
Hold
Click and hold the mouse button down to temporarily freeze the on screen display. Click-hold-drag down to keep
the display frozen, and down further to0 display earlier samples. Click again to “unfreeze” the display.

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30. Spectral Hold
The JS Spectral Hold plug-in can be used
to take a snapshot of what is playing,
then use a phase vocoder to hold that
sound until you tell it to stop The function
of the first four trigger button are:
0 – Update and hold
1 – Toggle hold (no update)
2 – Update
3 – Stop hold
They work by changing the settings in the
three drop down lists (hold, update
state, and clear hold), which are the last
three controls in the plug-in window. You
can also change these settings yourself,
directly, using these drop down lists.
Triggers 4 to 9 currently serve no
purpose. The other controls are:

Parameter Descriptions
FFT Size
Determines the FFT size used in the analysis. Seven discrete values between 512 and 32768 are available. Larger
FFT sizes provide probably better tones in the sound generated, but introduce more latency and use more CPU. In
any case, different sizes have very different sound.
Analysis overlap
Determines the overlap (as a function of FFT size) at which two samples are taken in order to determine what to
synthesize. Has no effect on performance.
Output overlap
Determines the overlap when synthesizing the held tone. Larger values increase CPU use. Smaller values can
have an audible pulsing effect.
Hold volume
Determines the level of volume at which output is held.
Dry mix during hold
Determines the level of volume of the dry (pre FX) signal in the mix when output is held.
Dry mix when not holding
Determines the level of the volume of the dry (pre FX) mix when output is not held.
Phase increase
Determines the effect to which the phase will be increased when a sound is held.
Hold drop-down
A drop down with three settings – off, hold, and hold (do not update on switch). Changing the hold state from
"off" to "hold" means a copy of the input audio is analyzed, stored, and immediately played.
Update state drop-down
A drop down with two settings – off (represented by - ) and update. Changing update state from "-" to update
means a copy of the input audio is analyzed and stored. If the Hold state is not in Off then the stored audio will
be played immediately. Otherwise, if the Hold state is switched from Off to “Hold (do not update...", then it will
switch to the stored audio.
Transport startbehavior
A drop down with two settings – clear hold and preserve hold.
Mix-in on Update
Determines the extent (range 0.0 to 1.0) to which the sound will be mixed in when update is applied.
Auto-update every …
A range of 0.0 to 30.0 seconds at which auto-updating can be set.

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31. Spectrograph Meter

The JS Spectrograph Meter is a spectrogram. It uses color to represent the frequencies in an audio signal as they
vary thru time. Its controls are as follows:

Parameter Descriptions
Scroll
Click to enable/disable this option. When enabled, the display will scroll from right to left as thetrack is played.
FFT size
The FFT size settings determines which algorithm is used to compute the display. A higher FFT size will give you
better frequency resolution. This can be preferred when you are trying to identify low-frequency sounds such as
plosives or mic handling noise. A lower FFT setting will give you better time resolution and is more useful when
needing to identify high frequency events, or when working with transient, percussive signals.
The default setting is 1024. Click and drag up or down to change this.
Windowing function
You have a choice of which windowing function you want applied in performing FFT analysis. These are blackman-
harris, blackman, rectangular or hamming. The default is blackman-harris.
Rate
Determines the speed at which the spectograph display progresses. Right click for a menu range of 1 to 8.where 1
is the slowest and 8 is the fastest. If you require faster speeds, click and drag up over the word rate for an
available range of 1 to 40.
Curve
This setting determines at which frequency a curve will be plotted. Adjusting this will control whether frequencies
are displayed linearly (curve=0), or if more pixels will be allocated to lower frequencies
Gate
Sets the floor of the noise gate. Amplitudes of frequencies under this value will be drawn as black.
Scroll, FFT size, Window and Rate can also be modified from the context menu – right click over any part of
the display.

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32. Super8 MIDI Controlled Synchronizer
Super8 is an eight channel loop sampler. You can record/play each channel independently, however when played
together, the channels will always play back in sync with one another.
Here is an overview. Follow the links at the
bottom of this page for more information.
Super8 has 8 outputs (1 for each channel),
and a 9th which is the monitoring output.
You can use this to set a separate
monitoring mix for the selected channel
(which can be useful for live performance).
When you activate record for a channel
and no other channels are active
(recording or playing), you are
simultaneously recording to that channel
as well as defining the loop length.
When you hit record or playback for that
channel again, it ends the loop and
immediately starts over - if you end the
loop with "play", it starts only playing the
loop you just recorded. If you end the loop
with record, it starts overdubbing it.
Some pointers to get you started:
 Each of the eight slots can record
audio (not MIDI). Recording into the
slots can be triggered from the GUI
or from a controlling MIDI device.
 The first slot that you record into
sets the length for all of the slots.
 Each channel has a number of controls, which are mappable to MIDI notes but also can be clicked on via
the mouse:
 In the top right of each slot is the monitoring control. This has 3 modes: auto-monitoring (meaning it only
monitors when this track is recording or focused). monitoring-on (always monitors), and monitoring off
(never monitors: this can be used for example for live drums etc).
 On the left: reverse. this reverses the channel's sample
 Record button - this starts recording on the channel. If the channel is already playing back or recording, it
sets overdubbing, otherwise it clears the channel before recording.
 Play button - this stops recording if recording, and starts playback. If the channel is already playing back,
it stops the channel.

To download a sample project that uses the Super 8 MIDI Controlled Synchronizer, follow this link:
forum.cockos.com/showthread.php?t=169085
The above thread also contains a link to an instructional video, and more useful information.
See also https://forum.cockos.com/showthread.php?t=169085 for more helpful tips.
For Kenny’s awesome video see: https://www.youtube.com/watch?v=hiSTeVlinAg

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