This quickstart manual is made to give you the quickest possible
introduction and overview to the functions of splinetime. To go into
further detail, read the corresponding Item Reference
the passage about the underlying audio engine
. More information can be found
in the splinetime
First, open the plugin within your effect chain and start playback in
As splinetime technically is an 'audioeffect', the plugin will start
playback when the host
plays (you can bypass this by pressing triggermode
In this case, playback will start when you click on an audiobuffer
Now, select one of the 16 audiobuffers
bottom-left, then fill it by either
. In the big display above you can see the content of the currently
selected buffer, and the yellow 'playback cursor'. If you are in a live
situation and want to process the incoming audio, leave recfree
activated, set the speed
faders to 1x and dont touch
The four areas to the right represent the four different dimensions of
audio processing : time
, and freq
Every single fader within an area sets the value at a specific
sync-time, just like a step-sequencer. Determine the 'duration' of one
fader with the unit
fader to the right (1/16
recommended, or 'static' to stay on the first fader). Then set dens
to maximum and env
to minimum, you can change
First concentrate on one effect area at a time to understand the usage.
Operating at all 4 dimensions simultaneously will soon lead you to the
experimental stage, and results in more funky and squashed sounds
(which can be quite nice, too).
So, check out the speed
alter the playback speed of the audio buffer down to zero and hear what
happens. I urge
you to also later check out the 'pure position' setting in the option
which lets you directly determine the playback position with the speed
area works the same way
and is quite
self-explaining - it alters the pitch. The freq
lowpass filter with adjustable resonance
Using the time
area is a bit more
approaching. For effective usage you need to take care of the value of
fader, it sets the length
of one 'sync unit', and with the time
faders you now can set how many 'events' should happen within this
unit. Perhaps describe a rising line with the faders, so you can in a
accelerate your sound. The time area is most productive if you set dens
to a value below 2, resulting in audible gaps. Alternatively, choose
'linked to dens' in the option
Finally, take a look at the options -
menu where you can set
playback behaviour. Decide if you want the buffer to run in an endless
loop, or to stop at the buffer or syncstep end.
This is it for the basics. Have fun finetuning your settings!
To understand the creation of sounds in splinetime, it can be helpful
understand the underlying audio-process. It is based on a granular
synthesis mechanism, which means that the source material in the
audiobuffer is permanently split up into small pieces (called 'grains')
and then, on the output side, reconstructed according to the settings
of the faders. Internally, there are at least four important values :
- is the size of one
split-up piece of music and lies in the
range of about 10 milliseconds to one second (determined by
The grainsize gets smaller with increasing values of the time
fader, because more events
( = grains) must fit into one 'sync duration'. In splinetime, the
by the size of the yellow cursor rectangle within the big buffer
determines the proceeding of the playback
'cursor' - eg. if the speed
fader is set to 0, the process will constantly play back at the same
position resulting in repetition of the same grain. If it is set to 1,
it will play back at the original speed of the audio source.
- according to the pitch
faders, the currently played grain is up- or downsampled to alter its
pitch, while the overall playback duration is not affected.
- every grain
will be faded in and out for smoothening the
sound. This is directly driven by the env
fader of splinetime.
With the granular technique, it is possible to alter the speed of audio
material without altering its pitch. There are several different
implementations of this technique available. The granular engine
used in splinetime
is a performance friendly algorithm which makes use of the upcoming
artefacts for artistic approaches.
click on a section to view the
Almost every fader movement
and even buffer
changes can be recorded and automated by your host sequencer, except
those which would not make sense to automate, like load / save
operations or the option
and file operations
Preset banks can be stored in .sxb files. See loadbank
for more information.
The buffer area consists of
16 seperate audiobuffers that can be
filled with audio data. The currently selected buffer is highlighted
and displayed in more detail in the big buffer window above. Clicking
on a buffer will restart playback. The small fader below each
buffer sets the start point within the audiobuffer. When
switching from one buffer to another, a sound morphing
between the two
buffers will take
place. See the option
for morph settings and other settings like customable fader sizes or
Each one of the 16 faders represent a value at a specific sync time.
Its duration is set
with the unit
fader. The time fader value itself
is responsible for how many events (grains, see granular engine
) will be played within
this duration, eg. within 1/16 note. For example, with the fader set to
minimum, there will be 1 event per fader unit. The resulting sound
depends heavily on the value of the dens
fader, for it creates gaps between the events. The fader behaviour can
be set in the option
The fader settings in the speed area determine the playback speed in
the audiobuffer and is visualized by the yellow cursor in the big
buffer display. While it's based on a granular
, the side artifacts like phasiness and comb filtering depend on the
values in the time
area and the unit
fader. The fader range can be set in the option
On the right side of the speed area are 5 small green buttons with
which all of the 16 speed faders can be set at once.
Note that the latency-free processing of the incoming audio by having
all the time, is only
possible with the speed faders set exactly to 1x speed. Elseway the
playback and recording cursor will not always be at the same position
and will run out of sync.
The fader settings in this area determine the current pitch of the
audio output, where the center fader setting represents 'original
pitch'. The fader range can be set in the option
menu. Due to performance reasons, the pitch algorithm used within
produce 'aliasing', that means that the typical pitch artifacts -
additional frequencies - can appear.
On the right side of the pitch area are 5 small green buttons with
which all of the 16 pitch faders can be set at once.
The fader values in this area represent the cutoff frequency of a
IIR lowpass filter with adjustable resonance
The value range is adjustable under the option
Displays the audio output
graphically. See options - scopeview
for further details.
Clicking on the display brings the 'About & Credit Box' to
Click on the box to remove it again.
Each instance of a splinetime
plugin can hold 16 presets (called
'programs') in memory. One program includes all settings like amp
fader values, the activated buffer
on. It does not include the settings within the option
menu, as these are globally fixed for all 16
Switching between programs is possible with the horizontal program
fader, or by pressing 'prev
for the previous or 'next
are used for copying one
program setting to another.
All 16 programs can be saved in one .sxb file via savebank
This fader is used to smooth the small slices of audio that are
generated in the underlying audio process by applying a fadein /
amplitude envelope. The type of envelope can be set in the option
Small values lead to a long attack and release time and to a softer
sound, but can result in amplitude modulation or audible gaps. This can
be compensated with the dens
The fader values from 1/1 to 1/64 note set the duration of one fader
Additionally there are the settings : 'static' -
step-sequencing and remains on the first fader - and 'bufsized' which
calculates the correct unit-duration for the length of the
buffer size. With the speed-faders set to
original speed (+1.0), the playback of the whole buffer will take
exactly the time of one sync-run-through (over all 16 faders).
Adjusts the resonance of the
filter used within the freq
The density of the resulting audio stream. A value of 1 or greater
constant output, while a value smaller than 1 results in gaps.
Be aware that the env
influences the occurrence of gaps. If you really want constant output,
set 'dens' to 2.0 and 'env' to 0.
In this menu, overall settings can be made. The current settings apply
to all 16 programs
within one instance
of the plugin. They can be stored inside a .sxb file via the savebank
To import options formerly saved in a .sxb file, select 'import
options'. This will not change parameter settings or audio buffer
time fader values
Selecting 'fixed values
(whole notes and
triplets) ', the time
fader values are quantised
to an exact number of events per fader unit, from 1 to 32.
Selecting 'free values' means variable fader
'fixed - linked to density' and 'free - linked to
density' result in the same fader behaviour like above,
but additionally the dens
value will change according
to the time fader values. With this
setting it is possible to move between a 'gapped' and a continous
speed fader range
Sets the speed
from 0 to 1x speed,
from 0 to 2x speed or
from - 2x (backward playback)
to 2x speed.
The 'pure position' settings disables speed
selection (playback speed is always 1). The speed
faders now determine the playback position within the currently
pitch fader range
fader range with octave or semitone resolution. The center setting of
the pitch fader is always at the original pitch.
filter cutoff frequency
cutoff frequency maximum position of the freq
fader. As this is a lowpass filter, a better resolution in the bass
range can be achieved when selecting a lower cutoff frequency like 5,5
: envelope type
windowing type used to smooth the small slices of audio that are
genrated in the underlying audio process.
The 'trapezoid' window is the simplest, a linear
fadein / fadeout. Useful especially when a constant output is
intended, with dens
set to 2x and env
set to minimum.
The 'raised cosine' window has a sinuslike fadein /
fadeout signal form. Setting env
minimum results in the same window like the classic 'hanning' window,
which can be selected, too, and where the env fader has
Switching the envelope type to 'off / rectangular',
attaches no window, and therefore results in a fast and 'hard' attack
audio output. The env fader has no effect.
: playback mode
mode' the audio playback will continously go on, while in the other
modes playback will end when the cursor reaches either the audiobuffer
or the end of the step-sequencing line, or both. If one of these modes
is selected, the playback cursor jumps back to the buffer start
that can be set with the small faders below each buffer
. According to the triggermode
setting, playback will stop or
: morph algorithm
algorithm, that is being called when switching between two
audiobuffers, can be
Each one has a different 'sound' and its result varies depending on
the audio content.
Note that the morphing algorithms are more
cpu-hungry, they use fft analysis and can therefore lead to cpu usage
peaks. Select 'off' for cpu-friendly, direct buffer switching without
: morph stepsize
the number of morphing steps between the two audiobuffers.
: scope view mode
audio visualisation in the small window at the bottom of splinetime can
set here. Either time domain (waveform view) or frequency domain
(spectral view, up to 10 kHz) can be selected, and, in addition,
whether the active, single grain
should be analyzed or the output sum.
Please note that the frequency domain view demands additional
calculations because of the Fourier Transformation that it makes use
of. This can lead to higher cpu usage. Under cpu-critical circumstances
it is better to switch it off.
: record sync unit
recording audio by pressing recsync
currently selected audiobuffer will be erased and resized according to
sync unit value in this menu.
The first entry in this menu loads a .sxb preset bank file, which
includes the settings for all 16 programs
, the audio content of all 16 audiobuffers
, and the
settings of the option
Importing the fader settings and parameters only, without
overwriting the audiobuffer contents, is possible by answering
the upcoming message box ('overwrite existing data...') with 'no'.
The other entries in the menu allow a quick access to the recent sxb
files that were opened. The filename references are stored in a file
called 'sxblist.txt' inside the vstplugins-directory or, if internally
not available, in the 'C:' directory.
Saves a .sxb preset bank file to disk, which includes the settings for
all 16 programs
, the audio
content of all 16 audiobuffers
, and the
settings of the option
menu. Note that a sxb file can get big in size because of the audio
data that is saved in it.
When saving the overall
host-sequencer song, splinetime's parameter settings are
too, but without the pure audio data. Instead the wav file references
are stored, so that everything gets restored when reloading the song.
With 'triggermode' activated, the audio processing will not restart
until an audiobuffer is clicked with the mouse or by midi / automation
control. It then jumps back to the buffer start position, that can be
set with the small faders below each buffer
Whether audio processing should be stopped at all can be set under options - playback mode
'triggermode' can also be used to override the 'general playback halt'
when the host sequencer is not playing.
Displays a graphic quantisation grid for the main faders.
With this button switched on, the incoming audio of
the mixer channel that reaches splinetime is being monitored.
Starts recording the incoming audio of the mixer channel as soon as the
host-sequencer playback reaches the next bar. The audiobuffer length
set in the option
menu. Recording stops
automatically after this period.
This button is useful if you rely on the time-sync context of your
music, eg. in live situations or for rythmic material.
You can monitor the incoming audio by clicking mon
Permanent recording of the incoming audio of the mixer channel into the
currently active audiobuffer. Monitoring whats currently coming in is
. Stop recording by pressing 'recfree' again.
this button to get your source
material, or use it as a 'live input' if all of the speed
faders are set to 1. In this case, the imaginary 'recording cursor' is
the point of the 'playback cursor'.
The buffer length can be set in the option
menu if this buffer is empty. If not, the existing audio will be simply
Opens a file selector for
loading an audio file into the currently
selected buffer. At the moment, only 16bit, 44.1kHz mono or stereo .wav
files are supported. The maximum size of audio data is limited to 20
seconds to keep memory and disk space usage low.
Opens a file selector to save
the content of the currently selected
audiobuffer to a 16bit, 44.1kHz stereo .wav file. Note that this is no
mixdown function, no effects are applied and only the pure audiobuffer
content will be saved. Use it to save live recorded stuff.
Erases the currently selected audiobuffer
Sets the output volume.
and graphics by Sebastian
FFT algorithm by Laurent de Soras : FFTReal
VST is a trademark of Steinberg Media Technologies GmbH
Thanks to the supportive vst community, see vstcode