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In the above menu In is the name of the QuickBus you are connecting to. You dont want to create a new QuickBus, you want to connect to one that already exists, and thats what youre doing. This is how your structure should look now:
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Instead of a nasty looking diagonal wire, we get two nice references, stating that the input and output are connected by a QuickBus whose name is In. Now we can go back out to the primary level and modify our structure to use the new lter weve just built. The Add and A/E modules can be thrown away. This is our nal result:
Takes quite a bit more CPU, doesnt it? Well, dont forget that this lter is modulated at audio rate in pitch scale. If you dont like it, you can still revert to the old structure or use the Multi 2-pole FM lter module from the primary level (slow envelopes, remember?), but we hope that you do like it. Even if you dont, there are quite a few other lters with new features that you might like better. And, if you dont like the new Reaktor Core lters, there are a whole bunch of other Reaktor Core modules you can try.
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2.3. Audio and control signals
Before we proceed we need to take a look at one particular convention used in the Standard Macros of the Reaktor Core library. The modules you nd in that area are best described in terms of several different types of signals: audio, control, event, and logic. We will explain event and logic signals a little bit later; for now well concentrate on the rst two types. Audio signals are obviously signals which carry audio information. These include signals taken at the outputs of oscillators, lters, ampliers, delays, and so on. Furthermore, modules such as lters, ampliers, saturators, delays and the like would normally receive an incoming audio signal to process. Control signals, on the other hand, do not carry audio, they are used to control other modules. For example, outputs of envelopes and LFOs as well as keyboard pitch and velocity signals do not carry any sound, but can be used to control a lters cutoff or resonance, or a delay lines delay time, and so on. Correspondingly, a lters cutoff or resonance input port, or a delays time input port are intended to receive control signals. Here is an example of a Reaktor Core lter module which you already know:
The upper input of the lter is for the audio signal to be ltered and, therefore, expects an audio-type signal. The F and Res inputs are obviously control type. The outputs of the lter carry different kinds of ltered audio, so all those signals are also audio type. A sine oscillator module, on the other hand, has only a single control input (for the frequency), and a single audio output:
The signal at the second input of this module will be attenuated according to the amount given at the A input and mixed with the signal at the chain (>>) input. The signal at the chain input is not attenuated. Such ampliers can be used to build mixing chains, where the >> port connections constitute a mixing bus:
In our case we dont need a mixing bus, but we can use this module to replace both our Audio Mix and Amount modules. The fed back signal will be attenuated by the amount specied by the Fbk input and mixed to the input signal exactly as it was before:
Congratulations, you have built a simple digital-echo effect. The next step is to add some tape feel to it.
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2.4. Building your rst Reaktor Core macros
In the echo effect we just built, we used a Delay 4p macro from the library, which gives us a reasonably high-quality digital delay. But, high-quality or not, it still sounds too digital. We will make it sound warmer by adding two features found in tape delays: saturation and utter. Lets start by deleting the delay macro from the structure and creating an empty macro instead. Right-click on the background as select Built-In Module > Macro:
Double-click it to dive inside. You will see an empty structure, similar to the one you are diving from:
It also works similarly, but there are some important differences because the previous one was a structure of a Reaktor Core cell, whereas this one is an internal structure of a Reaktor Core macro. These differences have to do with the available input and output modules, which are different:
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The Latch and Bool C types of ports will be explained much later in this manual and are used for advanced stuff. We are interested now only in the rst type, which is called Out (or In for inputs). Its a general type of port that can accept audio-, control-, event-, and logic-type signals. In fact, the port doesnt care whether its audio, control, event, or logic; the difference is important only for you as a user, because it describes how the signal is to be used; for Reaktor Core they are all the same. There is also no difference between audio/event inputs/outputs as on the previous structure level, because we dont have Reaktor primary-level signals on the outside any longer, it is pure Reaktor Core now. The rst thing we are going to do is name the macro, which is done in the same way as for core cells, by right-clicking on the background, selecting Owner Properties, and typing in the name:
The F input denes the rate of the gate repetitions, and the W input denes the duration of open gates (at 0 they are 50% of the gate period, at 1 its 0%, and at 1 its 100%). The Rst input restarts the LFO in response to incoming events (hence the LFO is restarted each time theres a gate event at the main gate input). The module connected to the Rst input of the Rect LFO is called Value and can be found in Standard Macro > Event Processing. It ensures the LFO is restarted at zero phase by replacing the values of all incoming events by the value at its lower input, which is zero. The LFO output is converted into a gate signal by using a Ctl2Gate converter, also found in Standard Macro >Event Processing. Remember, LFOs do not work inside event core cells. If you want to try out this structure, youll need to use an audio core cell. REAKTOR CORE 55
3. Reaktor Core fundamentals: the core signal model
3.1. Values
Most of the outputs of Reaktor Core modules produce values. (Producing a value means that at any moment in time there is a value associated with the output.) The values are available to all modules whose inputs are connected to those outputs. In the following example an adder module gets values 2 and 3 from the two modules whose outputs are connected to its inputs, and it produces a value of 5 at its output.
If you want to draw an analogy to the hardware world you can think of values as signal levels (voltages), especially with relatively large-scale modules such as oscillators, lters, envelopes, and so on. However, values are not limited to those kinds of processingthey are just values and can be used to implement any processing algorithm, not just voltage-modeling algorithms.
3.2. Events
Time is not continuous in the digital world; it is discrete. Probably the most familiar example of this is that a digitally stored recording doesnt store the full information about an audio signal, which is continuously changing over time, but rather stores only information about the signal level at regularly spaced points in time. The number of points per second bears the famous name of sampling rate.
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Here is a picture of a continuous signal:
and its digital representation:
Because we are in the digital world, the outputs of our modules cannot change values continuously. On the other hand, we dont have to limit ourselves to changing values at regularly spaced points in time. For one thing, we do not have to maintain a particular sampling rate all over our structures. For another thing, in certain areas of our structures we do not even have to maintain any sampling rate at all; that is, our changes do not have to happen at regular intervals. For example, at time zero the output of our adder could have a value of 5. The rst change could occur at time 1 ms (one millisecond). The second change could occur at 4 ms. The third at 6 ms:
If macros and built-in modules are the same then nothing should change when we replace the multiplier by a Z^-1 macro:
But it is different, because the implicit feedback is now gone. There must be something special about the Z^-1 macro. And, in fact, there is. If we look inside this macro well see almost the same structure as the one we mentioned earlier to implement the Z^-1 functionality: 94 REAKTOR CORE
As you can see, the clock input of the macro is connected to the internal Read module. The default connection for this input is not to a zero constant, but the audio clock, and thats what you would want in most cases. The module connected between the upper input and the write module will be explained later, for now just ignore it. So far, theres nothing special about this macro, except that it seems to implement the Z^-1 structure we have discussed earlier. So how does the Reaktor Core engine know that this structure is meant to resolve feedback loops? Obviously, the engine can know that it can resolve feedback loops, but how does it know that its intended to? This is controlled by the Solid setting in the macro properties:
The Solid property tells the Reaktor Core engine whether the macro is to be considered as a solid built-in module for the purposes of feedback resolution or whether it is to be considered transparent. In 99% of the cases, you would want to keep this property on. Thats because you typically dont want implicit feedback resolution to happen inside your macros. One reason for that is that the resolution happening inside a macro wont be visible unless you go into the macro, so that some of the implicit feedback REAKTOR CORE 95
delays can go unnoticed. For example, we can take our previous structure with the Thru macro and disable the Solid setting (make sure you are editing the Solid setting for the right macro, you can see it by the Thru text in the label eld of the properties):
Make sure youre editing the right macro
Disable the Solid setting
Now your outside structure probably still looks the same (we say probably because you never can be sure where exactly the automatic feedback resolution will happen):
But if you change your structure a little, connecting the output to another module, it could look like this:
Our feedback resolution delay seems gone. So in a larger and more complicated structure we could easily miss the fact that theres an implicit delay. Wheres this delay gone? Of course, its now inside the Thru macrothe only place 96 REAKTOR CORE
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The available criteria are: = equal != not equal () <= less or equal () < less >= greater or equal () > greater It is, of course, possible to connect several routers to the same comparison module, in which case they will change their state simultaneously. The Router module splits the event path into two branches. Quite often these branches will later be merged:
Depending on the result of the comparison the above structure will either invert the input signal or leave it intact. An alternative implementation of this structure would be:
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In this version, the 0 output of the Router is disconnected; therefore, the Router works as a gate, letting the events through only if its in the true state. The inverted value then arrives at the second input of the Merge, thus overriding the non-inverted value, which is always arriving at the rst input. If the router is in false state the inverter doesnt receive an event and doesnt send an event to the second input of the Merge; therefore, the original unmodied signal goes to the output of the Merge. The branches are most often merged with a Merge module. But theoretically speaking you could use many other modules (for example, arithmetic modules like adder, multiplier, and so on) instead. Routers treat the initialization event just like any other event. Therefore, one could lter out the initialization event by using routers, thereby ensuring that the initialization event wont appear in particular areas of the structure.
6.2. Building a signal clipper
Lets build a Reaktor Core macro structure that would clip the incoming audio signal from the top at a specied level:
If the input signal is not greater than the threshold it will be routed to output 0 of the Router and, through the Merge, to the output of the structure. Otherwise, the signal will be routed to output 1, where it triggers the latch, sending the threshold value to the Merge instead. The same thing happens during initialization. Note that this structure will not change its output in response to changes to the threshold. Rather the new threshold value will be used for the next and all subsequent events at the signal input. This is in a way similar to a modulation macros behavior, where modulator changes do not result in output events.
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Switching between oat and integer types (if its supported by the module) is done in the Signal Type property of the module:
A module set to integer type will process the input values as integers and produce integer output values. You can tell that a module is in integer state by the fact that its signal inputs and outputs look different:
There is no such thing as default signal type for macros. The reason is that normally you wouldnt build structures that process integers in exactly the same way as structures processing oats and vice versa (although you might for some relatively simple structures). Integer signals can be freely interconnected with oats, but the wires created between different type signals will perform signal conversion, which can use a certain amount of CPU. At the time of this writing, the extra CPU usage is somewhat noticeable on PCs and quite signicant on Macs. The OBC connections of oat and integer types are not compatible with each other, of course. There can also be information loss during such conversions. In particular, large integers cannot be precisely represented by oats, and obviously, oats cannot be precisely represented by integers. Large oats (larger than the largest representable integer) cannot be represented as integers at all, in which case the result of the conversion in undened. During oat-to-integer conversion, the values will be rounded approximately to the nearest integer. We say approximately because the result of rounding 0.5 can be either 0 or 1, although you can rely on the fact that 0.49 will be rounded to 0, and 0.51 to 1. REAKTOR CORE 113
It is important to understand that turning the processing mode of an operation to integer and converting of a floating point result of the same operation to an integer is not the same. Lets consider an example. Here we are adding two numbers 2.4 and 4.3 as floats. The result is clearly 6.7, which when converted to integer will produce 7. So the output of the following structure is 8:
Now if we change the mode of the first adder to integer, instead of adding 2.4 and 4.3 we will add their rounded versions which are 2 and 4 respectively, producing 6. So the result is 7:
Clock inputs completely ignore their incoming values, therefore they are normally always oats. Furthermore, signal type conversion will not be performed for the signals that are used only as clocks:
Here the clock input of the Read module is still oat although the module has been set to integer mode (the OBC ports look the same regardless whether they are oat or integer). Integer feedback is automatically resolved in the same way as oat feedback by inserting an integer mode Z^-1 module (of course no denormal canceling is needed here).
Obviously, the playback position must be the Distance in samples behind the record position; therefore, we subtract one from the other:
The distance value is latched because it is produced by a control signal input, which potentially can receive events at any time, and we do not want the subtraction happening at times other than at audio-clock events. If we just subtract, the difference can turn out to be less than zero because our array is not a loop; its ends are not connected together. So we need to wrap the result: -1 must become N-1, -2 must become N-2, -3 must become N-3, and so on.
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So we include another macro for wrapping:
Because we know that the difference cannot be smaller than N+1 (because RecordPos is between 0 and N-1 and Distance is between 0 and N-1), wrapping can be implemented as simple addition of N:
Lets get back to our top level structure. Now that we have the write and read indices, we just need to perform reading and writing:
Note that reading is happening after writing and that its clocked by the sampling-rate clock. Heres a proposed test structure. Dont forget to put an ms2sec converter into the Delay core cell and to set the Delay core cell to monophonic mode:
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Actually its a good idea to switch the delay to monophonic mode as soon as possible, because each voice will consume about 200K of memory. 44,100 samples, using 4 bytes (32 bit) each: 44,100*4 = 176,400 bytes, which is a little bit more than 172K (a kilobyte has 1024 bytes). To test the above structure play notes on your midi keyboard and hear them delayed by the amount of time specied by the Time knob.
8.4. Tables
Theres another module similar to Array. The name of this module is Table and it can be found in the Built-In Module > Memory submenu:
The difference between a table and an array is that you can only read from a table; you cannot write to it. The values in a table are pre-initialized using the modules properties. To get access to the list of the values press the button in the properties window:
Produces a BoolCtl signal at the output indicating the result of comparison of the input values. The value at the upper input is placed to the left of the comparison sign and the value at the lower input to the right (so that the module on the picture above checks if upper value is greater than the lower one). PROPERTIES: Criterion the comparison operation to be performed
F.19. Flow > Compare Sign
Produces a BoolCtl signal at the output indicating the result of the sign comparison of the input values. The value at the upper input is placed to the left of the comparison sign and the value at the lower input to the right (so 154 REAKTOR CORE
that the module on the picture above checks if the sign of the upper value is greater than the sign of the lower one). The sign comparison is dened as follows: + is equal to + is equal to + is larger than The sign of zero value is undened, so arbitrary result may be produced should one of the compared values be zero. PROPERTIES: Criterion the comparison operation to be performed
F.20. Flow > ES Ctl
Produces a BoolCtl signal at the output indicating the momentary presence of an event at the input (that is, the control signal is true if there is an event at the input of this module at the given moment).
F.21. Flow > ~BoolCtl
Produces a BoolCtl signal at the output which is an inversion of the input BoolCtl signal (true changes to false and vice versa).
F.22. Flow > Merge
An output event is sent each time there is an event at any of the inputs or at several of them simultaneously. If only one input receives the event at a given time, the value of the output event will be equal to the value of the input event. If several inputs receive an event simultaneously, the value at the lowest input (among those receiving the event) will be selected. For example, if both second and third (counting from top) inputs receive an event, the value at the third input will be taken. PROPERTIES: Input Count number of inputs of the module
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F.23. Flow > EvtMerge
The functionality is similar to that of the Merge module, except that all input values are ignored. The value of the output event is undened. This module is intended to be used for generating signals to be used as clocks. Works only in oating point mode, since the value is not meant to be used anyway. PROPERTIES: Input Count number of inputs of the module
F.30. Memory > Table
Denes a pre-initialized read-only array. The module itself does not perform any action. All operations on the table are to be performed by the modules connected to the table output which is an OBC slave connection of array type. PROPERTIES: FP Precision edit the values in the table controls the formal precision of the output connection
F.31. Macro
Provides a container for an internal structure. The number of inputs and outputs is not xed and is dened by the internal structure. PROPERTIES: FP Precision controls the formal precision of the output connection Look changes between Large (label and port names visible) and Small (label and port names invisible) looks Pin Alignment controls the alignment of the ports in the outside view of the macro Solid controls the treatment of the macro by the core engine. If turned off the macro boundary is transparent for feedback resolution and possibly other things. Leave it ON unless you really, really know what youre doing! Icon button loads a new icon for the macro, clears the icon (no icon assigned) button
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Appendix G. Expert macros
G.1. Clipping > Clip Max / IClip Max
The signal at the upper input is clipped from the top by the threshold value at the lower input. Changes to the threshold do not generate events.
G.2. Clipping > Clip Min / IClip Min
The signal at the upper input is clipped from the bottom by the threshold value at the lower input. Changes to the threshold do not generate events.
G.3. Clipping > Clip MinMax / IClipMinMax
The signal at the upper input is clipped from the bottom by the threshold value at the middle input and from the top by the threshold value at the lower input. Changes to the thresholds do not generate events.
G.4. Math > 1 div x
Computes the reciprocal of the input value
G.5. Math > 1 wrap
Wraps the incoming value into the range [-0.5.0.5] (the wrapping period is 1).
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G.6. Math > Imod
Computes the remainder of the division of upper value by the lower value. The output event is sent each time there is an event at either of the inputs or at both of them simultaneously.
H.18. Audio Shaper > 3-1-2 Shaper
Audio signal shaper with variable amount of 2nd and 3rd order distortion. The REAKTOR CORE 169
distortion amount and type is controlled by the Shp input: Shp = 0 no shaping Shp > 0 3rd order shaping Shp < 0 2nd order shaping
H.19. Audio Shaper > Broken Par Sat
Broken parabolic saturator. Has a linear segment around the zero level. L input species the output level for the full saturation (typical value = 1). H input species the hardness (range 01). Larger values correspond to a larger linear segment in the middle. S input controls the symmetry of the shaping curve (range 11). At 0 the curve is symmetric.
H.20. Audio Shaper > Hyperbol Sat
Simple hyperbolic saturator. The L input species the full saturation output level (default = 1). However the full saturation is never reached with this type of saturator.
H.21. Audio Shaper > Parabol Sat
Simple parabolic saturator. The L input species the full saturation output level (default = 1). Note: the full saturation is reached at the input level equal to 2L.
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H.22. Audio Shaper > Sine Shaper 4 / 8
4th / 8th order sine shaper. The 8th order shaper has a better sine approximation but takes more CPU.
H.23. Control > Ctl Amount
Linear invertible control of the amount (amplitude) of the control signal. A=0 turns off the signal A=1 leaves the signal unchanged A = -1 inverts the signal Typical usage: controlling modulation amount
H.24. Control > Ctl Amp Mod
Modulates the control signals amplitude by a given amount (AM) in linear scale. AM = 1 doubles the amplitude AM = 0 no change AM = -1 mutes the signal
H.25. Control > Ctl Bi2Uni
Changes a 11 bipolar signal into a unipolar one. The a input controls the amount of change, at 0 theres no change, at 1 there is 100% change (default is 1). Typical usage: connect immediately after an LFO to adjust the polarity of the modulation. REAKTOR CORE 171
H.40. Convert > sec2Hz
Converts time period in seconds into corresponding frequency in Hz. E.g. 0.1sec 10 Hz.
H.41. Delay > 2 / 4 Tap Delay 4p
2/4-tap delay with 4 point interpolation. T1T4 inputs specify the delay time in seconds for each of the taps. The maximum delay time defaults to 44,100 samples which is 1sec at 44.1kHz. To adjust the time change the size of the array in the delay macro.
H.42. Delay > Delay 1p / 2p / 4p
1-point (non-interpolated)/2-point interpolated/4-point interpolated delay. T input species the delay time in seconds. The maximum delay time defaults to 44100 samples which is 1sec at 44.1kHz. To adjust the time change the size of the array in the delay macro.
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Use interpolated versions of delays for modulated delays. For non-modulated (xed time) delays non-interpolated version is normally better.
H.43. Delay > Diff Delay 1p / 2p / 4p
1-point (non-interpolated)/2-point interpolated/4-point interpolated diffusion delay. T input species the delay time in seconds. The Dffs input sets the diffusion factor. The maximum delay time defaults to 44,100 samples which is 1sec at 44.1kHz. To adjust the time change the size of the array in the delay macro.
H.44. Envelope > ADSR
Generates an ADSR Envelope. A, D, R specify attack, decay and release times in seconds S species sustain level (range 0.1, at 1 sustain level is equal to the peak level) G gate input. Positive incoming events (re-)start the envelope. Zero or negative events close the envelope GS gate sensitivity. At zero sensitivity the envelope peak has always amplitude of 1. At sensitivity equal to one, the peak level is equal to the positive gate level. RM retrigger mode. Selects between analog/digital mode and between retrigger/legato mode. In digital mode the envelope always restarts from zero while in analog mode the envelope restarts from its current output level. In retrigger mode consecutive positive gate events will restart the envelope, while in legato mode it restarts only when the gate changes from negative/zero 176 REAKTOR CORE
RM RM RM RM
= = = =
to positive. The allowed RM values are following: analog retrigger (default) analog legato digital retrigger digital legato
H.45. Envelope > Env Follower
Outputs a control signal which follows the envelope of the incoming audio signal. The A and D inputs specify the follow attack and decay time parameters in seconds.
H.46. Envelope > Peak Detector
Outputs the last peak level of the incoming audio as a control signal. The D input species the output level decay time parameter in seconds.
Performs a disjunction of two logical signals (the output is 1 if at least one of the inputs is 1). For input values other than 0 or 1 the result is undened.
H.95. Logic > XOR
Performs an exclusive disjunction of two logical signals (the output is 1 if one of the inputs is equal to 1 and the other equal to 0). For input values other than 0 or 1 the result is undened.
H.96. Logic > Schmitt Trigger
Switches the output to 1 if the input value becomes larger than L+ (default 0.67), switches the output to 0 if the input value becomes less than L- (default 0.33).
H.97. Oscillators > 4-Wave Mst
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Generates 4 phase-locked audio waveforms. The frequency is specied by the F input (in Hz). The pulse width is specied by the pw input (range 1.0.1, affects only the pulse waveform). This oscillator can oscillate at negative frequencies and additionally offers a synchronization output for 4-Wave Slv oscillator.
H.98. Oscillators > 4-Wave Slv
Generates 4 phase-locked audio waveforms. The frequency is specied by the F input (in Hz). The pulse width is specied by the pw input (range 1.0.1, affects only the pulse waveform). This oscillator can oscillate at negative frequencies and can be synchronized to another 4-Wave Mst/Slv oscillator. The SncH input controls the synchronization hardness (0 = no sync, 1 = hard sync, 01 = various degrees of soft sync). A synchronization output for another 4-Wave Slv oscillator is also provided.
H.99. Oscillators > Binary Noise
Binary white noise generator. Outputs randomly alternating values of 1 and 1. An incoming event at the Seed input would (re-)initialize the internal random generator with a given seed value.
H.100. Oscillators > Digital Noise
Digital white noise generator. Outputs random values in the range 1.1 An incoming event at the Seed input would (re-)initialize the internal random generator with a given seed value.
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H.101. Oscillators > FM Op
Classical FM operator. Outputs a sine wave whose frequency is dened by the F input (in Hz). The sine can be phase-modulated by the PhM input (in radians). An incoming event at the Rst input would restart the oscillator to the phase specied by the value of this event (range 0.1).
H.102. Oscillators > Formant Osc
Generates a waveform with a fundamental frequency specied by the F input (in Hz) and the formant frequency specied by the Fmt input (in Hz).
H.103. Oscillators > MultiWave Osc
Generates 4 phase-locked audio waveforms. The frequency is specied by the F input (in Hz). The pulse width is specied by the pw input (range 1.0.1, affects only the pulse waveform). This oscillator cannot oscillate at negative frequencies.
H.104. Oscillators > Par Osc
Generates a parabolic audio waveform. The F input species the frequency in Hz.
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H.105. Oscillators > Quad Osc
Generates a pair of phase-locked sine waveforms with a phase shift of 90 degrees. The F input species the frequency in Hz.
H.106. Oscillators > Sin Osc
Generates a sine wave. The F input species the frequency in Hz.
H.107. Oscillators > Sub Osc 4
Generates 4 phase-locked subharmonics. The fundamental frequency is specied by the F input (in Hz). The subharmonic numbers are specied by S1. S4 inputs (range 1.120). The Tbr input controls the harmonic content of the output waveform (range 0.1).
H.108. VCF > 2 Pole SV
2-pole state-variable lter. The F input species the cutoff in Hz and the Res input species the resonance (range 0.0.98). The HP/BP/LP outputs produce highpass, bandpass and lowpass signals respectively.
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H.109. VCF > 2 Pole SV C
2-pole state-variable lter (compensated version). Offers an improved behavior at high cutoff settings. The F input species the cutoff in Hz and the Res input species the resonance (range 0.0.98). You also can use negative resonance values which will smear the slope further. The HP/BP/LP outputs produce highpass, bandpass and lowpass signals respectively.
H.110. VCF > 2 Pole SV (x3) S
2-pole state-variable lter with optional oversampling (x3 version) and saturation. The F input species the cutoff in Hz, the Res input species the resonance (range 0.1), the Sat input species the saturation level (typical range 8.32). The HP/BP/LP outputs produce highpass, bandpass and lowpass signals respectively.
H.111. VCF > 2 Pole SV T (S)
2-pole state-variable lter with table compensation and optional saturation (S version). Offers an improved behavior at high cutoff settings, but slightly different from the 2 Pole SV C version. The F input species the cutoff in Hz, the Res input species the resonance (range 0.1), the Sat input species the saturation level (typical range 8.32). The HP/BP/LP outputs produce highpass, bandpass and lowpass signals respectively. 192 REAKTOR CORE

Getting Started Guide
Disclaimer
The information in this document is subject to change without notice and does not repre sent a commitment on the part of Native Instruments GmbH. The software described by this document is subject to a License Agreement and may not be copied to other media. No part of this publication may be copied, reproduced or otherwise transmitted or record ed, for any purpose, without prior written permission by Native Instruments GmbH, herein after referred to as Native Instruments. All product and company names are or trade marks of their respective owners. Document authored by: Aleksander Rebane Product Version: 5.5 (06/2010) Document version: 1.0 (06/2010) Special thanks to the Beta Test Team, who were invaluable not just in tracking down bugs, but in making this a better product.
Contact
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Table of Contents
Welcome to REAKTOR 1.1 1.2 1.3 1.4 1.5 2.1
System Requirements REAKTOR Modes: Full, Player and Demo How to Get Started The REAKTOR Documentation Special Formatting used in this Document Configuring the Audio Hardware 2.1.1 Accessing the Audio and MIDI Settings dialog 2.1.2 Selecting an Audio Hardware Device 2.1.3 Selecting a Sample Rate 2.1.4 Adjusting Output Latency 2.1.5 Routing: Assigning REAKTOR's Inputs 2.1.6 Routing: Assigning REAKTOR's Outputs Configuring the MIDI Hardware A Few Important Buttons Ensembles, Instruments and KOMPLETE Instruments Loading Carbon 2 3.3.1 Opening the Sidepane 3.3.2 Opening the Browser 3.3.3 Loading a File Playing Carbon 2 and its Snapshots 3.4.1 Loading a Snapshot Adjusting the Sound 3.5.1 Switching Effects
Basic Settings in REAKTOR
2.2 3.1 3.2 3.3
Loading and Playing
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3.6 4.1 4.2 4.3
3.5.2 Changing the Filter Movement 3.5.3 Changing the Filter Settings Saving Your Settings Recap Using the Browser to Load the Ensemble Instruments inside Ensembles 4.3.1 Junatik Panel Overview 4.3.2 The Difference Between an Ensemble and an Instrument Snapshot Master for Plug-In and Snapshot Hierarchies 4.4.1 Snapshot Hierarchy as seen in Properties 4.4.2 Recall by MIDI and Snapshot Master for Plug-In 4.4.3 Snapshot Properties for the Junatik Instrument 4.4.4 Snapshot with a REAKTOR Switch Turned Off 4.4.5 Snapshot Flexibility Swapping the Delay Instrument for a Reverb Instrument 4.5.1 Opening a REAKTOR Structure 4.5.2 Removing an Instrument from the Structure Using the Bowser to Search for Instruments Connecting the Instrument to the Audio Out 4.7.1 Making the Connections 4.7.2 Resetting the Positions of Instruments in the Ensemble Panel Restoring the Correct Snapshot Hierarchy 4.8.1 Opening the Instrument Properties Recap and Overview Creating the Synth Sequencer Ensemble 5.2.1 Create a New Ensemble
Modifying a REAKTOR Ensemble
4.6 4.7
Creating a Sequenced Synthesizer 5.1 5.2
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5.5 5.6
5.2.2 Using the Browser to Load the Instruments Making the Synth Sequencer Work 5.3.1 Overview of the Structure 5.3.2 Removing the Unwanted Instrument 5.3.3 Connecting ANALOG Outputs to the Audio Out 5.3.4 Overview of Monoliner and ANALOG Making the MIDI Connections between Instruments 5.4.1 Using Internal MIDI Connections REAKTOR Run and Stop Buttons Looking More Closely at Monoliner 5.6.1 Changing the Direction and Speed 5.6.2 Number of Steps, Step Offset and Shuffle/Swing 5.6.3 GATE, VELOCITY and PITCH Knobs and On/Off Buttons Exploring the Snapshots/Show Hints 5.7.1 Exploring Snapshots in Monoliner and ANALOG 5.7.2 Show/Hide Hints Recap and Overview Using the Browser to Launch Memory Drum 2 Replacing and Editing Samples in the Sample Map Editor 6.3.1 Opening the Map and Playing the Samples 6.3.2 Using Replace in the Edit Sample List 6.3.3 Sample Key-Split and Root Note 6.3.4 Using Add from the Edit Sample List 6.3.5 Editing the Key-Split The Memory Drum Interface 6.4.1 Changing Pitch for Just one Sample Slot 6.4.2 Exploring the Memory Drum Snapshots
All of these documents are available from REAKTORs Help menu.
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Special Formatting used in this Document
In addition to the manuals coming with REAKTOR, there is more information available on line: Be sure to visit the Native Instruments user forum at http://www.native-instruments.com/forum/. The huge community of REAKTOR users is sharing tips and tricks, and will help you with specific questions.
This manual uses particular formatting to point out special facts and to warn you of poten tial issues. The icons introducing these notes let you see what kind of information is to be expected:
Whenever this exclamation mark icon appears, you should read the corresponding note carefully and follow the instructions and hints given there if applicable.
This lightbulb icon indicates that a note contains useful extra information. This informa tion may often help you to solve a task more efficiently, but does not necessarily apply to the setup or operating system you are using. However, it should be worth reading for most users.
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REAKTOR 5 is capable of running as a stand-alone software with its own interface to your audio and MIDI hardware. This way you can play REAKTOR using a MIDI keyboard/control ler attached to your computer. Hereafter we will assume that you are running REAKTOR as a stand-alone application. In this chapter you will learn how to link REAKTOR to the MIDI devices and audio hard ware connected to you computer. You will also get to know REAKTOR a little bit without the distraction of making music using your DAW. Before you start your work with REAKTOR 5 it is a good idea to configure its audio settings to fit your needs. This only applies to the stand-alone version of REAKTOR 5, as with the REAKTOR 5 plugins all things audio and MIDI are handled by the host software.
Configuring the Audio Hardware
In order to listen to the sound REAKTOR generates, you need to configure your audio hard ware device (i.e., your soundcard or external audio interface) for use with REAKTOR.
Use Low-Latency Drivers Whenever possible you should use low-latency drivers while working with REAKTOR. RE AKTOR works with two types of low-latency drivers: ASIO Core Audio (only on computers running MacOSX) These technologies have been developed to ensure an efficient data transfer between soft ware and audio hardware and should provide a latency that is suitable for live play, if not unnoticeable.
Accessing the Audio and MIDI Settings dialog
When you start REAKTOR for the first time, the Audio and MIDI Settingsdialog should open automatically. You can call up this dialog again at a later point in time from within REAKTOR: 1. Click the Menu button to open the application menu.
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Select the File > Audio and MIDI Settings menu entry:
Fig. 3.7 The Browser tab
This makes the Sidepane show the File Browser. It is split into three parts: At the top, you have four buttons to access four different types of content. Below, there is a folder tree. And at the bottom, there is a list of files contained within the selected folder from the tree above. Click the Player button at the Browsers top. You will see all installed KOMPLETE Instru ments listed in the upper area. Select the package REAKTOR Factory Selection with a mouse click.
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Fig. 3.1 The Browser with Factory Selection selected
The button next to the Player button switches its name depending on your activation of REAK TOR: It holds the factory content of REAKTOR and reads Factory after REAKTOR is activat ed. Until then, it is labeled Demo. This indicates that REAKTOR will switch into Demo mode if you load any Ensemble from that location.
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Playing Carbon 2 and its Snapshots
Loading a File
As you selected the REAKTOR Factory Selection, its contained instruments are listed in the Browsers lower area. Click the entry Carbon2 and drag it to the applications main area. REAKTOR now loads the file.
Fig. 3.2 Carbon 2 loaded in REAKTOR PLAYER
Carbon 2 is a classic subtractive synthesizer with sections for oscillators, filters, modula tion sources and integrated effect units.
Check the Instrument Reference for a detailed explanation of Carbon 2, or the documentation that comes with REAKTOR Factory Selection.
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Loading a Snapshot
Play some notes on your MIDI keyboard to get an idea of how the synthesizer sounds. Then, lets change the sound completely by loading a different Snapshot. A Snapshot is REAKTORs terminology for a sound, preset, or patch. Each Instrument can hold Snapshots, and loading any of these Snapshots will set each control of that Instru ment to a specific value, thus re-creating a particular sound. The Snapshots of Carbon 2 are accessible from the central control in the applications Main Bar.
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Fig. 3.3 The Snapshot control
1. 2. 3.
Click into the Snapshot control to open a drop-down menu. The menu holds all Snap shots of the instrument. Select the entry In Deep. Play some notes to hear the difference.
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Adjusting the Sound
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Please refer to your hosts documentation for details about storing host projects.
To store the Ensemble file as a copy of the original instrument, select the Save As entry from the application menu. This will open a default window to select folder and filename. After you selected the new location and typed in a new filename, click the Save button.
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In this tutorial: We will modify an existing REAKTOR Ensemble called Junatik. Junatik comprises a Synth Instrument and a Delay Instrument. We will insert a Reverb Instrument into the Structure, and connect it to the Synth. We will remove the Delay Instrument from the Structure. We will also learn more about REAKTORs Snapshot system. We will use the Browser to load up Junatik, but also to insert the Reverb Instrument.
If you are using REAKTOR PLAYER, you do not need to continue reading. After you have read chapter 2 and 3 you are set to go!
In the first tutorial we learnt how to create and save Snapshots using the Append button. Also we saved our work, using Save Ensemble Ascommand. As we will be using these techniques again, you might want to have a look at the first tutorial again before proceed ing. Using REAKTOR Buttons To recap, in the last tutorial we used: Click the Sidepane button in the Main Bar to open and close the Sidepane. The Side pane is where the Browser, Snapshot, Panelsets, and Properties tabs are located.
Fig. 4.1 The Sidepane button
Clicking the Browser tab opens the Browser.
Fig. 4.2 The Browser tab
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Using the Browser to Load the Ensemble
Clicking the Snapshot tab opens the Snapshots list.
Fig. 4.3 The Snapshot tab
In this tutorial we will learn about REAKTOR Properties. We will also learn about the Structure button. Clicking the Properties tab opens the Properties of REAKTOR Ensembles, Instruments, Macros, Modules, et cetera.
Fig. 4.4 The Properties tab
Clicking the Structure button opens REAKTORs Structure.
Fig. 4.5 The Structure button
It is a good idea to get used to these buttons and their icons as they are very useful, espe cially later on when you use REAKTOR as a plug-in!
Again we will use the Browser to load the Ensemble. 1. Click on the Sidepane button to open the Sidepane. 2. Click on the Browser tab to open the Browser. 3. Click on the Factory button to navigate to the Factory Library. 4. Using the folder tree in the top part of the Browser, navigate to Ensembles > Classics > Synthesizer. You should now see a list of Synths in the bottom part of the Browser. 5. In the bottom part of the Browser, double-click on Junatik to open it.
Click on the Function button to see the Function page of Junatiks Properties.
Fig. 4.12 The Function button
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Fig. 4.13 Ensemble Properties with Recall by MIDI and Snapshot Master ticked
Have a look above at the Properties, in the Snapshot area, the checkboxes Recall by MIDI and Snapshot Master for Plug-In, are both engaged.
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Recall by MIDI and Snapshot Master for Plug-In
Snapshot Master for PlugIn means that when these Snapshots are changed by the En semble (the master), the Snapshots for the Synth and the Delay will also change to gether. In other words, the Junatik Instrument and StereoTDelay Instrument are slaves to the Ensemble master. Recall by MIDI means that this Instrument can receive Program Change messages. If the master also has this setting and is on the same MIDI channel, the master's Snap shot with take precedence.
Snapshot Properties for the Junatik Instrument
If you single click on the Junatik Panel, you will see that the Recall by Parent box is ticked.
Fig. 4.14 The Snapshot area of the Junatik Instrument Properties with the Recall by Parent checkbox engaged
This confirms that the Snapshots on the Ensemble Panel, are the Master Snapshots. Try changing some Snapshots and you will see how different Snapshots in the Junatik In strument, sometimes (but not always), have different Snapshots in the StereoTDelay In strument. Lets choose Snapshot 24 Pointer 1. Note that the StereoTDelay Snapshot says, 1 Delay off. Now try choosing different Snapshots in the Junatik Instrument.
Snapshot with a REAKTOR Switch Turned Off
You will notice that the Snapshots in the Junatik Instrument all play back with no Delay effect: Snapshot 1 in the StereoTDelay has the Delay turned off, because the On switch above the Wet knob, is turned off.
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Swapping the Delay Instrument for a Reverb Instrument
Fig. 4.15 The Delay is switched off
Switches like this not only Mute the Instrument, they also conserve CPU resources. So if you used this Ensemble in a DAW, and used the DAWs Effects, it would make sense to turn the switch off.
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Exploring a REAKTOR Sampler
In this tutorial we will have a look at Memory Drum 2 which is a very useful sample player for drum kit samples.
Now we have explored two of the Synths in the REAKTOR Library, it is time to have a look at one of its Sampler Players. We will learn how to replace one the samples in the Sample Map with one of own samples. We will also learn how add a sample to the Sample Map. We will look at how to select samples in the Sample Map using our MIDI keyboard, and how we can audition samples by using our mouse. We will also look at Sample Mapping in REAKTOR. Finally, we will have a brief look at some of the unique features of Memory Drum. So first, we are again going to use the Browser to load up Memory Drum 2.
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Using the Browser to Launch Memory Drum 2
Fig. 6.1 Memory Drum 2
In the Sidepane, click on the Browser tab to open the Browser.
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In the folder tree, navigate to Ensembles > New Additions > Sample Player. Double-click on MemoryDrum2.ens to launch it.
Fig. 6.2 Choosing Memory Drum 2 in the Browser
Memory Drum 2 is mapped so that C1 will play the first sample and each semitone will play a new sound. First of all, explore the sounds by playing different pitches on your key board. Maybe you are happy with the samples but you would like to replace them with some of your Samples? To do that, we need to open the Sample Map Editor.
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Replacing and Editing Samples in the Sample Map Editor
Now we will use the Sample Map Editor to replace Samples. We will open the Sample Map, use SEL BY KEY so our MIDI keyboard can play Samples in the Map, and also use our mouse to audition Samples.
Opening the Map and Playing the Samples
Double-click on the Sample waveform. The Sample Map Editor will open up. Alterna tively you can also press the Sample Map button in the Sidebar or use the keyboard shortcut [F9].
Turn on the Select Sample by Key feature by pressing the Key button in the top-right of the Sample Map. You can now select Samples with your MIDI keyboard.
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Click on the Sample Prehear button to hear the samples in their original form.
Fig. 6.3 The Sample Prehear button
Lets replace some samples with some of our own ones. Lets replace the snare sample on MIDI Note 48. First of all we need to select it by pressing C2 on our keyboard.
Inserting Pitch and Gate Modules
Next we need to add a couple of componentsModules in REAKTOR terminology. The Note Pitch and Gate Modules will enable our MIDI keyboard to play this Synth. 1. Right-click if you are in Windows ([Ctrl]+click if you are on Mac OS X) on an emp ty space in the Instrument Structure and choose the Built-In Module > MIDI In > Note Pitch menu entry to insert the Note Pitch Module into the Structure.
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Repeat the same procedure but choose the Built-In Module > MIDI In > Gate menu entry for the Gate Module.
Connecting the Pitch and Gate Modules
Now we need to make connections so that we can play our fledgling Synth. 1. Using the same click and drag techniques as before, connect the NotePitch Module to the P (Pitch) Input of the Osc 3 Wave Macro. 2. Then connect the Gate Module to the G (Gate) Input of the ADSR-Env Macro.
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Inserting the Filter
Finally, we need to connect the top Out from the ADSR-Env to the A (Amplitude) of the Osc 3 Wave Macro. Now your Synth should look like this:
Fig. 7.12 Your Synth with finished connections
Safe Output Level
Before we play our Synth, we need to check that the Output Level will be safe. 1. To do this, first go to the Ensemble Panel by clicking on the Panel button.
Look at the Master Instruments Panel, and reset the Level Fader to -10.
Fig. 7.13 Use the Master Instrument to select a safe output level
Now play a few notes, and you should hear the Oscillator change pitch and start and stop with your key presses!
The next thing to add is a Filter Macro.
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Inserting the Filter Macro and Doing Some Rewiring
Make sure you are inside the Instrument Structure. If you are looking at the Panel, first press the Structure button.
Then double-click the Instrument Object to enter its Structure. Right-click if you are on Windows ([Ctrl]+click if you are on Mac OS X) on a blank part in the Instrument Structure and choose the Macro > Building Blocks > Filters > 4 Pole Filter (BP, BLP, LP) menu entry.
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Make a Sampler Using REAKTOR Modules
We will now build a Sampler using REAKTOR Modules.
We have already explored samples with the REAKTOR Sampler Memory Drum 2. We have also build a Synth using ready-made REAKTOR Macros. Following on from this, we are going onto build a Sampler from scratch using REAK TOR Modules.
What Kind of Sampler?
We will build a Sampler that is ideal for playing back vocal loops, drone textures and sound effects, etc. The samples will be able to playback forward, reversed and forward/backwards. We will use REAKTORs basic Sampler Module. We will use an Envelope and a Filter Mod ule, and connect the controls such as Attack, Decay, Sustain and Release for the Enve lope, and Cutoff and Resonance for the Filter.
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Building the Basic Structure
Fig. 8.1 The Finished Sampler
We will start in much the same way as when we built our Synth. So we need to start off by creating a New Ensemble.
Load up a New Ensemble
Press the Menu button and choose the File > New Ensemble menu entry.
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Fig. 8.2 Creating a new Ensemble
On this occasion we will use the default Stereo In/Stereo Out Instrument. This gives us the option of changing our Sampler at a later stage to one with left and right outputs.
Inserting the First Modules
We will now insert the Sampler Module along with Pitch and Gate Modules. 1. Go to the Ensemble Structure by clicking on the Structure button. 2. If you remember, double-clicking on the Instrument in the Ensemble Structure will open it, ready for inserting some Modules. 3. In Windows, right-click (in Mac OS X [Ctrl]+click) on a space inside the Instru ment Structure and choose the Built-In Module > Sampler > Sampler menu entry.
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Next insert the NotePitch and Gate Modules. If you remember, we did this when we built our Synth. So choose the Built-In Module > MIDI In > NotePitch menu entry and then from the same place in the Structure context menu the Built-In Module > MIDI In > Gate menu entry. Your Structure should look something like this:
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Inserting the Envelope
Next we will insert an ADSR Envelope and controls for Attack, Decay, Sustain and Re lease. In Windows, right-click (in Mac OS X [Ctrl]+click) on a space inside the Instrument Structure and choose the Built-In Module > LFO, Envelope > ADSR menu entry.
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Fig. 8.3 Inserting an ADSR Module
Adding the Controls for the Envelope
We will use the same technique to create Controls that we used when we added Faders to the Mixer in our Synth. 1. To add an Attack Fader to the Envelope we need to right-click in Windows ([Ctrl] +click in Mac OS X) on the A or red dot beside it, and choose the Create Control menu entry.
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2. Repeat this process for the D, S, and R Inputs of the Envelope. Now our Structure should look like this:
Now we need to connect the Sampler to our Outputs, and the Gate and NotePitch Modules need to be connected to the Sampler and ADSR-Env, using click / hold and drag.
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Connecting the Sampler to the 2 Audio Voice Combiners
Click on the Out dot of the Sampler, hold the mouse button down, and drag the wire to left Audio Voice Combiner. 2. When you have reached the black dot In of the Audio Voice Combiner, let go of your mouse button. The wire should connect. 3. Repeat the same procedure for the right Audio Voice Combiner. Again, the yellow lamp confirms the connection!
Fig. 8.4 Sampler connected to Audio Voice Combiners
Connecting the NotePitch and Gate Modules
Click/hold on the red dot of the NotePitch Module, and drag the wire to the P (Pitch) Input of the Sampler. 2. Let go when you touch the red dot. 3. Repeat the same procedure for the Gate Module, which needs to be connected to the Trig Input of the Sampler and the G (Gate) Input of the ADSR-Env. Your Structure will now look something like this:
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Loading a Sample into the Sampler
We have built a simple Sampler, but now we need to load a sample, just to hear that it works. First thing to do is to open the Sample Map Editor.
Opening the Sample Map Editor from the View Menu
Click on the Sample Map button in the Sidebar.
Fig. 8.5 The Sample Map button
You will see the Sample Map Editor below the Instrument.
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Now we have the Sample Map Editor, we do the same procedure as before when we added a sample to the Sample Map in Memory Drum 2.
Adding a Sample to the Sample Map Editor
Changing the Settings of the Existing Sample
Before we add more samples, we need to make some changes to the existing sample. The idea is that we will map our samples to the white MIDI Keys starting on MIDI Note 48. So first of all we need to make sure that our sample will playback correctly on MIDI Note 48. We also want the sample to just play back on that note.
Making Changes to the Key-Split and Root Key
We want our sample to playback at its correct pitch when MIDI Note 48 is played. The first thing to change is the Root Notecurrently at 60to 48. Double-click the entry in the Root column and type in the new value. Since we only want the sample to playback on that key, you need to activate the Sin gle Key Mode. To do this, select the Single Key Mode menu entry from the Edit dropdown menu.
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Now you can change the L and R Key-Split mapping to 48. Highlight a Sample, dou ble-click on one of the entries in the corresponding row and columns (L or R) and then type in the new value. Press [Enter] on your computer keyboard when you are finished. The sample will only play back on MIDI Note 48 at its correct Pitch.
Fig. 8.10 Remapping the Sample
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Sample Reverse and Other Playback Possibilities
Adding More Samples
will add another sample and map the new sample to MIDI Note 50. Again use the Add menu entry from the Edit drop-down menu to add another sample. Now map that Sample so the L and R Key-Split is also set to 50. We want to reproduce this procedure, and add two more samples. We want to keep them on the white keys, so the next sample should have L and R and Root set to 52. 4. The fourth sample should have L and R and Root set to 53. Now our 4 samples will play back using the notes C2, D2, E2, and F2that is MIDI Notes 48, 50, 52, and 53. Our Sample Map should look something like this:
We 1. 2. 3.
Fig. 8.11 Four samples mapped to single keys.
If you like, you can add more samples on MIDI Notes 55, 57, 59, and 60. But for the purpose of this tutorial, we will move on to explore sample playback. We will learn how to change the direction of sample playback.
Now we will explore ways to loop reverse and transpose samples.
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Looping, and Direction Buttons
This might be a simple Sampler, but each sample can have some different characteristics in terms of how it plays back. Lets have a closer look at some of the features in the Sam ple Map Editor. Below the sample waveform you can see Buttons for Loop, Reverse, and Alternating Loop.
Fig. 8.12 Buttons and checkboxes for Loop, Reverse, and Alternating Loop
As you can see the defaults for these 3 features are off. 1. Make sure the Key button is active so we can use our MIDI keyboard to select a sam ple. Lets start off with the one on MIDI Note 48.
Now click on the Loop buttonits background will light up.
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Use your MIDI keyboard to play the note corresponding to the sample you are editing and keep holding the note down. You should hear the sample looping by going back to its beginning and then repeating. Now engage the Reverse checkbox.
Now you should hear the Sample looping but this time reversed. Engaging the Alternate checkbox will cause the sample to be played back reversed and then forwards.
Try other combinations! Each of your 4 Samples can have different settings.
You could also place the same sample over 4 keys and have different playback characteristics!
Transposing a Sample
At the moment all 4 of our samples are playing at their original pitch. Use the Root settings to change the Pitch. In my example, I have transposed the sample down 7 semitones by changing the Root note to 7 semitones higher.
Fig. 8.13 Transposing a Sample
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Adding a Filter
Now we will add a Filter to our Sampler, and create controls for Cutoff and Resonance. We will also try different filter characteristics such as High Pass, Band Pass, and Low Pass.
Inserting the Filter Into the Structure
In Windows, right-click (in Mac OS X [Ctrl]+click) on a space inside the Instrument Structure and choose the Built-In Module > Filter > Multi 2-Pole menu entry.
Fig. 8.14 Inserting a Filter.
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Creating the Filter Controls and Making the Connections
We are going to create controls for the P (Pitch Cutoff) Input and the Res (Resonance) In put. 1. As before, right-click on the respective Inputs if you are in Windows ([Ctrl]+click if you are in Mac OS X) and choose the Create Control menu entry.
Now connect the output of the Sampler to the Input of the Multi 2-Pole, and the Out put of the LP (Lowpass) Output of the Filter to the 2 Audio Voice Combiners. After moving the Modules around to create some space, your Structure should look some thing like this:
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After making both connection and moving the Selector, the Structure looks like this:
Fig. 8.22 Connections made
Finally we need to unlock the Instrument Panel to move the Fader, by clicking on the Panel Lock button and dragging the Pos Fader to the right.
Fig. 8.23 Panel Lock button
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Fig. 8.24 Moved Position Fader
Lock the Panel by pressing the Panel Lock button again.
8.10.7
Explore Your Sampler and Make Some Snapshots
Make some changes to the settings of your DIY Sampler and save them as Snapshots.
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Remember, we press the Snapshot tab and use the Append button to store our Snapshot.
Fig. 8.25 The Snapshot Tab
Fig. 8.26 Using Append
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As you know, you must save your Ensemble again to save your new Snapshots. You can again use the Save Ensemble command or if you decided you want to keep an earlier ver sion of the Sampler, maybe one without the Selector, then use the Save Ensemble As command and type in an amended name.
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Now that you have completed the tutorials in the REAKTOR Getting Started guide you can turn to the more advanced techniques by reading the other manuals that REAKTOR comes with. Or you can head directly to the REAKTOR user forums on our website for the latest tips and for a growing number of beautifully crafted Instruments and Ensembles. Wherever you go from here, have fun with REAKTOR and dont forget to share your crea tions with the user community!
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