Effects
-------

Effects are, generally, blackboxes that transform audio signals in a
specified way. More exactly, the only input data for an effect in ZynAddSubFX
is:

* an array of samples, which is read *on line*
* the current system time (used for LFOs)

The output is the transformed array of samples.

NOTE: As described, effects have no information about anything else. For
example, key presses are not recognized. Therefore, pressing a key does not
initiate the LFO. Phase knobs will always be relative to a *global* LFO, which is
only dependent on the system time.

ZynAddSubFX has 3 types of effects:

* System Effects
* Insertion Effects
* Instrument Effects

TODO: Describe these 3 types (their differences).

[[effects::general_topics, General Topics]]
General topics
~~~~~~~~~~~~~~

* *Wetness* determines the mix of the results of the effect and its input. This
mix is made the effects output. If an effect is wet, it means that nothing of
the input signal is bypassing the effect. If it is dry, then the effect has no
effect. TODO: Difference between Volume and D/W?
* *Pan* lets you apply panning, which means that the sound source can move to 
the right or left. Set it to 0.0 to only hear output on the right side, or to 
the maximum value to only hear output on the left side.
* *LRc.* or *L/R* let you apply crossover.
* *Filter stages* are the number of times that this filter is applied in series.
So, if this number is 1, you simply have this one filter. If it is two, the
sound first passes the filter, and the results then pass the same filter again.
In ZynAddSubFX, the wetness is applied after all stages were passed.
* *LFOs* are, as the name says, oscillators with, compared to the frequency of 
the sound, low frequency. They often appear in order to control the effect. 
They can have some of the following controls:
** *LFO Type* determines the shape of the LFO. If not present, the LFO is a 
sine wave.
** *Freq* determines the LFO's frequency.
** *Dpth* is a multiplier to the LFO. Thus, it determines the LFOs amplitude 
and its influence.
** *Rnd* is the LFO amplitude's randomness
** *St.df* lets you determine how much left and right LFO are phase shifted.
64.0 means stereo, higher values increase the right LFO relatively to the left
one.
********************************************************************
Hint: Keep in mind that Effects that can be controlled by LFO can also be 
controlled arbitrary: Set the LFO depth to zero and manipulate the phase knob
(e.g. with NRPNs or maybe via OSC in the future).
********************************************************************

Equalizer
~~~~~~~~~

Introduction
^^^^^^^^^^^^

An http://en.wikipedia.org/wiki/Filter_%28signal_processing%29[equalizer] is a
filter effect that applies different volume to different frequencies of the
input signal. This can, for example, be used to "filter out" unwanted 
frequencies.
ZynAddSubFX's implementations follow the
http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt["Cookbook formulae for
audio EQ"] by Robert Bristow-Johnson.

Filter Types
^^^^^^^^^^^^

This topic is completely discussed in <<filters, the Filters section>>.

Usage
^^^^^

We describe all parts of the GUI here. The term passband (or often just "band")
refers to the amount of frequencies which are not significantly attenuated by
the filter.

* *Gain* (on the left) defines an offset which is added to the complete filter.
* *B.* lets you choose the passband number. Multiple passbands define one
filter. This is important if you want multiple filters to be called after each
other. Note that filters are commutative.
* *T.* lets you choose the current filter's type, as described above.
* *Freq* describes the frequencies where the filter has its poles. For some
filters, this is called the "cutoff" frequency. Note, however, that a bandpass
filter has two cutoff frequencies.
* *Gain* is only active for some filters and sets the amount of a special peak
these filters have. Note that for those filters, using the predefined gain makes
them effectless.
* *Resonance* lets you describe a peak at the given frequency for filters with
2 poles. This can be compared to real physical objects that have more gain at
their resonance frequency.
* *St.* lets you define multiple filter stages. This is equivalent to having
multiple copies of the same filter in sequence.

Chorus
~~~~~~

Introduction
^^^^^^^^^^^^

In a chorus, many people sing together. Even if each of them sings at exactly
the same frequency, all their voices usually sound different. We say they have a
different timbre. Timbre is the way we perceive sound and makes us differ
between different music instruments. This is, physically, achieved by varying
both the amplitude envelope and the frequency spectrum. Multiple sounds with
slightly different timbres make a sound more shimmering, or powerful. This is
called the chorus effect.

Function
^^^^^^^^

The chorus effect can be achieved by multiple people singing together. In
a concert, there are many instruments, resulting in the same effect. When making
electronic music, we only have an input wave and need to generate these
different timbres by ourselves. ZynAddSubFX therefore simply plays the sound,
pitch modulated by an LFO, and adds this to the original sound. This explains
the diagram below: The multiple pitches are generated by a delayed version of
the input. This version is being pitched by an LFO. More detailled, this pitch
is generated by varying the reading speed of the delayed sound; the variation
amount is controlled by an LFO.

image:./gen/chorus.png[width=700,
	title="The chorus effect. z^(-n.m) describes the delay."]

TODO: Add LFO pointing to delay?

Related effects to Chorus are Flangers. Flangers can be described as Chorus
with very short LFO delay and little LFO depth. You can imagine a flanger as two
copies of a sound playing at almost the same time. This leeds to interference,
which can be clearly heared. It is popular to apply flangers to guitars, giving
them more "character".

Usage
^^^^^

* First, crossover is applied.
* The following 5 knobs (*Freq*, *Rnd*, *LFO Type*, *St.df*, *Depth*) control 
the LFO for the pitch. If the depth is set to zero, the pitch will not be 
changed at all.
* *Delay* is the time that the delayed sound is delayed "on average". Note that
the delay also depends on the current pitch.
* After the correct element of the sound buffer is found using the LFO, the
*Fb* knob lets you set how loud it shall be played. This is mostly redundant to
the *D/W* knob, but we have not applied panning and substraction yet. 
* Next, the singal can be negated. If the *Substract* Checkbox is activated,
the amplitude is multiplied by -1.
* Finally, *Pan* lets you apply panning.

Distortion
~~~~~~~~~~

Introduction
^^^^^^^^^^^^

Distortion means, in general, altering a signal. Natural instruments
usually produce sine like waves. A wave is transformed in an unnatural way when
distortion is used. The most distorted waves are usually pulse waves. It is 
typical for distortion to add overtones to a sound. Distortion often increases 
the power and the http://en.wikipedia.org/wiki/Loudness[loudness] of a signal, 
while the db level is not increased. This is an important topic in the
http://en.wikipedia.org/wiki/Loudness_war[Loudness War].

NOTE: As distortion increases loudness, distorted music can cause ear damage
at lower volume levels. Thus, you might want to use it a bit careful.

Distortion can happen in many situations when working with audio. Often, this is
not wanted. In classical music, for example, distortion does not occur
naturally. However, distortion can also be a wanted effect. It is typical for
Rock guitars, but also present in electronic music, mostly in Dubstep and
DrumNBass.

The basic components of distortion are mainly

* a preamplifier
* the waveshaping function
* filters

Preamplification changes the volume before the wave is shaped, and is indeed the
amount of distortion. For example, if you clip a signal, the louder the input
gets, the more distortion you will get. This can have different meanings for
different types of distortions, as described below.

********************************************************************
The filters are practical. A reason for using them afterwards is that distortion
can lead to waves with undesired high frequency parts. Those can be filtered out
using the LPF. A reason for using filters before applying is to achieve
multiband distortion. ZynAddSubFX has no "real" multiband distortion by now,
however.
********************************************************************

Types of Distortion
^^^^^^^^^^^^^^^^^^^

This topic is completely discussed in
<<adsynth::oscilllator::types_of_waveshaping, the Oscillator Section>>. Note
that you can use the 
Oscillator editor in order to find out what your distortion effect does. Also 
note that while the Oscillator editor's distortion is limited to some 
oscillators you can produce in the Oscillator editor, the distortion effect can 
be used on every wave that you can generate with ZynAddSubFX.

Function
^^^^^^^^

We explain the functionality in a diagram and list the components below.

image:./gen/distort.png[width=700,
	title="The components of a distortion function."]
	
* Negation is the first thing to happen. If the *Neg* Checkbox is activated, the
amplitude is multiplied by -1.
* Panning is applied. Note, however, that you have to activate the
Stereo Checkbox, labeled *St*, before.
* Pre amplification is done next. The amount can be changed using the
*Drive* nob. Indeed, this is the amount of distortion. For example, if you clip
a signal, the louder the input gets, the more distortion you will get. This can
have different meanings for different types of distortion, as described above.
* *HPF* and *LPF* are filters with 2 poles. Whether they are used before or
after the waveshape, depends on the checkbox labeled *PF*.
* The next step is the wave shape. This defines how the wave is
actually modified. The *Type* ComboBox lets you define how. We will discuss some
types below.
* After the wave shape, we scale the level again. This is called
output amplification. You can change the value using the *Level* knob.
* Crossover is the last step. This is controlled by the knob *LR Mix* and
means that afterwards, a percentage of the left side is applied to the right
side, and, synchronously, the other way round. It is a kind of interpolation
between left and right. If you set the LR Mix to 0.0, you will always have a
stereo output.

Dynamic Filter
~~~~~~~~~~~~~

Introduction
^^^^^^^^^^^^

A dynamic filter is, as the name says, a filter which changes its parameters
dynamically, dependent on the input and current time. In ZynAddSubFX,
frequency is the only variable parameter. It can be used as an "envelope
following filter" (sometimes referenced "Auto Wah" or simply "envelope filter").

Function
^^^^^^^^

Though this filter might look a bit complicated, it is actually easy. We divide
the parameters into two classes:

* *Filter Parameters* are the ones you get when you click on *Filter*. They
give the filter its basic settings.
* *Effect Parameters* are the other ones that control how the filter changes.

The filter basically works like this: The input signal is passed through a
filter which dynamically changes its frequency. The frequency is an additive of:

* the filter's base frequency
* an LFO from the effect parameters
* the "amplitude" of the input wave

image:./gen/dynamic.png[width=700,
	title="The components of a dynamical filter"]

The amplitude of the input wave is not the current amplitude, but the so called
https://en.wikipedia.org/wiki/Root_mean_square["Root Mean Square (RMS)"] value.
This means that we build a mean on the current amplitude and the past values.
How much the new amplitude takes influence is determined by the *Amplitude
Smoothness* (see below).

********************************************************************
RMS value plays an important role in the term loudness. A fully distorted
signal can sound 20 db louder due to its higher RMS value. This filter takes
this into account, depending on the smoothness.
********************************************************************

Usage
^^^^^

* The 4 knobs in the middle (*Freq*, *Rnd*, *LFO Type*, *St.df*) control the
LFO.
* Two knobs let you control the way how the RMS value of the amplitudes is
measured:
** *A.M* sets the Amplitude Smoothness (this is described above). The higher
you set this value, the more slow will the filter react.
** *A.Inv.*, if being set, negates the (absolute) RMS value. This will lower
the filter frequency instead of increasing it. Note that this will not have
much effect if the effects input is not very loud.
* The following controls define the mix of the LFO and the amplitude.
** *A.S* sets the Amplitude Sensing (i.e. how much influence the amplitude
shall have).
** *LfoD* sets the LFO depth.
* The filter button lets you choose the filter type.
* After the input signal has passed through the filter, *Pan* can apply
panning.

Echo
~~~~

Introduction
^^^^^^^^^^^^

The echo effect, also known as
http://en.wikipedia.org/wiki/Delay_%28audio_effect%29[delay effect], simulates
the natural reflection of a sound. The listener can hear the sound multiple
times, usually decreasing in volume. Echos can be useful to fill empty parts of
your songs with.

Function
^^^^^^^^

In ZynAddSubFX, the echo is basically implemented as the addition of the
current sound and a delayed version of it. The delay is implemented as in the
picture below. First, we add the delayed signal to the effect input. Then,
they pass an LP1. This shall simulate the effect of dampening, which means that
low and especially high frequencies get lost earlier over distance than middle
frequencies do. Next, the sound is delayed, and then it will be output and added
to the input.

image:./gen/echo.png[width="700",
	title="The echo includes a fb line, labeled as z^-n, and a delay."]

********************************************************************
The exact formula in the source code for the dampening effect is as follows:

latexmath:[$Y(t) := (1-d) \cdot X(t) + d \cdot Y(t-1)$],

where latexmath:[t] be the time index for the input
buffer, latexmath:[d] be the dampening amount and latexmath:[X,Y] be the input,
respective the output of the dampening. This solves to

latexmath:[$Y(z) = Z(Y(t)) = (1-d) \cdot X(z) + d \cdot Y(z) \cdot z^{-1}$]

latexmath:[$\Leftrightarrow H(z) := \frac{Y(z)}{X(z)} = \frac{1-d}{1 -
d \cdot z^{-1}}$]

which is used in latexmath:[$Y(z) = H(z) \cdot X(z)$]. So latexmath:[$H(z)$] is
indeed a filter, and by looking at it, we see that it is an LP1. Note that
infinite looping for d=1 is impossible.
********************************************************************

Description
^^^^^^^^^^^

* *Pan* lets you apply panning of the input.
* *Delay* sets the time for one delay.
* *LRdl.* means Left-Right-Delay. If it is set to the middle, then both sides
are delayed equally. If not, then the left echo comes earlier and the right
echo comes (the same amount) later than the average echo; or the other way
round. Set the knob to 0 to hear on the right first.
* *LRc.* applies crossover.
* Feedback describes how much of the delay is added back to the input. Set
*Fb.* to the maximum to hear an infinite echo, or to the minimum to just
hear a single repeat.
* The *Damp* value lets the LP1 reject higher frequencies earlier if
increased.

Reverb
~~~~~~

Introduction
^^^^^^^^^^^^

A http://en.wikipedia.org/wiki/Reverberation[Reverberation] actually expresses
the effect of many echoes being played at the same time. This can happen in an
enclosed room, where the sound can be reflected in different angles. Also, in
nature, thunders approximate reverbs, because the sound is reflected in many
different ways, arriving at the listener at different times.

In music, reverbs are popular in many ways. Reverbs with large room size can be
used to emulate sounds like in live concerts. This is useful for voices, pads,
and hand claps. A small room size can simulate the sound board of string
instruments, like guitars or pianos.

Function
^^^^^^^^

As mentioned, a reverb consists of permanent echo. The reverb in ZynAddSubFX is
more complex than the echo. After the delaying, comb filters and then allpass
filters are being applied. These make the resulting sound more realistic. The
parameters for these filters depend on the roomsize. For details, consider the
information about https://ccrma.stanford.edu/~jos/pasp/Freeverb.html[Freeverb].

image:./gen/reverb.png[width=700,
	"The reverb, being similar to the echo."]

Description
^^^^^^^^^^^

* The *Type* ComboBox lets you select a reverb type:
** *Freeverb* is a preset. It was proposed by Jezar at Dreampoint.
** *Bandwidth* has the same parameters for the comb and allpass filters, but it
applies a unison before the LP/HP. The unison's bandwidth can be set using *bw*.
** Random chooses a random layout for comb and allpass each time the type or
the roomsize is being changed.
* The roomsize (*R.S.*) defines parameters only for the comb and allpass
filters.
* *Time* controls how long the whole reverb shall take, including how slow the
volume is decreased.
* The initial delay (*I.del*) is the time which the sounds need at least to
return to the user. The initial delay feedback (*I.delfb*) says how much of the
delayed sound is added to the input.
* Low pass filter (*LPF*) and high pass filter (*HPF*) can be applied before
the comb filters.
* The dampening control (*Damp*) currently only allows to damp low frequencies.
Its parameters are being used by the comb and allpass filters.
* *Pan* lets you apply panning. This is the last thing to happen.


Phaser
~~~~~~

Introduction
^^^^^^^^^^^^

The http://en.wikipedia.org/wiki/Phaser_%28effect%29[Phaser] is a special
dynamic filter. The result is a sweeping
sound, which is
often used on instruments with a large frequency band, like guitars or strings.
This makes it typical for genres like rock or funk, where it is often modulated
with a pedal, but also for giving strings a warm, relaxing character.

Function
^^^^^^^^

The audio signal is split into two paths. One path remains unchanged. The other
one is sent to a delay line. The delay time (the so called *phase*) is made
dependent on the frequency. Therefore, an all-pass filter is applied to the
signal, which *preserves* the amplitude, but determines the delay time. In the
end, both paths are added.

The following picture describes how this works on white noise. Light blue
signalises that the frequency is not present at the current time, and dark blue
signalises the opposite. The dark blue peaks appear if the delay time is very
short, because then, the second path almost equals the first one, which results
in duplication of the signal. If the delay line is very long, then it is --- in
the case of white noise --- totally at random whether the delayed signal
currently duplicates the unchanged path, or whether it cancels it out to zero.
This random effect results in white noise between the clear blue structures.

image:./images/phaser-spectrogram.jpg[width="700",
	title="Spectrogram of an 8-stage phaser
	modulated by a sine LFO applied to white noise."]

Phaser Types
^^^^^^^^^^^^

ZynAddSubFX offers different types of phasers:

* Analog and "normal" phasers. Analog phasers are more complicated. They sound
punchier, while normal phasers sound more fluently. However, analog filters
usually need more filter stages to reach a characteristic sound.
* Sine and triangle filters. Note that an analog triangle filter with many poles
is a barber pole filter and can be used to generate
http://en.wikipedia.org/wiki/Shepard_tone[Shepard Tones],
i.e. tones that seem to increase or decrease with time, but do not really.
* The LFO function can be squared. This converts the triangle wave into a hyper 
sine wave. The sine squared is simply a faster sine wave.
* TODO: Barber is deactivated, since PLFOtype is only 0 or 1?

Description
^^^^^^^^^^^

For the normal phaser, first, the LFO is generated:

* There are 4 controls (*Freq*,*Rnd*,*LFO tpye*,*St.df*) that define the 
LFO.
* *Phase* and *Depth* are applied afterwards in the usual way (TODO: I don't 
understand the code here for the normal phase...). For the analog phaser, 
*Phase* is not implemented, yet.
** If *hyp* is being set, then the LFO function is being squared.

Next, the input is being used.

* *Analog* decides whether the phaser is analog or "normal".
* First, *Pan* applies panning to the original input in every loop.
* Next, barber pole phasing is being applied (Analog only).
* *Fb* applies feedback. The last sound buffer element is (after 
phasing) multiplied by this value and then added to the current one. For normal 
filter, the value is added before, for analog after the first phasing stage.
* Now, *Stages* phasing stages are being applied. *dist* sets the distortion 
for when applying the phasing stages. This has only effect for analog phasers.
* The feedback is taken now.
* In the end, *Substract* inverts the signal, multiplying it by -1.

Alienwah
~~~~~~~~

Introduction
^^^^^^^^^^^^

The AlienWah effect is a special, dynamic
http://en.wikipedia.org/wiki/Formant[formant] filter (TODO: is this true?).
Paul Nasca named it AlienWah because it sounded "a bit like wahwah, but more
strange". The result of the filter is a sound varying between the 
vocals "Ahhhhh" (or "Uhhhhh") and "Eeeeee".

Function
^^^^^^^^

The way that the filter moves between the two vocals is mainly 
described by an LFO. A bit easified, Paul Nasca has stated the formula (for
latexmath:[$i^2=-1; R<1$]) as

latexmath:[$fb=R*(\cos(\alpha)+i*\sin(\alpha))$]

latexmath:[$y_n=y_{n-delay}*R*(\cos(\alpha)+i*\sin(\alpha))+x_n*(1-R)$].

The input latexmath:[$x_n$] has the real part of the samples from the wavefile
and the imaginary part is zero. The output of this effect is the real part of
latexmath:[$y_n$]. latexmath:[$\alpha$] is the phase.

Description
^^^^^^^^^^^

* *Pan*
* The following 5 controls (*Freq*,*Rnd*,*LFO tpye*,*St.df*, *Dpth*) define the 
LFO.
** *Fb*

** *Delay* If this value is low, the sound is turned more into a
"wah-wah"-effect.
** *Phase* See latexmath:[$\alpha$] in the above formula. This lets you set
where the vocal is between "Ahhhhh" and "Eeeeee".
** *L/R* applies crossover in the end of every stage. This is currently not 
implemented for the Analog Phaser.
