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 and the passage about the underlying audio engine . More information can be found in the splinetime forum .

First, open the plugin within your effect chain and start playback in your host-sequencer. 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 clicking loadwav , recsync or recfree . 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 them anymore!

The four areas to the right represent the four different dimensions of audio processing : time , speed , pitch , 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 it later.
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 area and 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 menu, which lets you directly determine the playback position with the speed faders.

The pitch area works the same way and is quite self-explaining - it alters the pitch. The freq area controls a 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 the unit 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 way 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 menu.

Finally, take a look at the options - playback menu where you can set the 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 to 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 :
grainsize - is the size of one split-up piece of music and lies in the range of  about 10 milliseconds to one second (determined by host tempo, time and unit faders).
The grainsize gets smaller with increasing values of the time fader, because more events ( = grains) must fit into one 'sync duration'. In splinetime, the grainsize is visualized by the size of the yellow cursor rectangle within the big buffer display.
playback speed -  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.
grain pitch - 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.
grain envelope - 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 derivatives and 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.



automation
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 settings.

presets and file operations
Preset banks can be stored in .sxb files. See loadbank and savebank for more information.


audiobuffer area
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 menu for morph settings and other settings like customable fader sizes or visualisation.

time area
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 menu.

speed area
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 engine , 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 menu.
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 activated recfree 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.

pitch area
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 splinetime can 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.

freq area
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 menu.

scope display
Displays the audio output graphically. See options - scopeview for further details.
Clicking on the display brings the 'About & Credit Box' to front. Click on the box to remove it again.

program area
Each instance of a splinetime plugin can hold 16 presets (called 'programs') in memory. One program includes all settings like amp , dens , time fader values, the activated buffer and so on. It does not include the settings within the option menu, as these are globally fixed for all 16 programs.
Switching between programs is possible with the horizontal program fader, or by pressing 'prev' for the previous or 'next' for the next program.
'copy' and 'paste' are used for copying one program setting to another.
All 16 programs can be saved in one .sxb file via savebank.



env
This fader is used to smooth the small slices of audio that are generated in the underlying audio process by applying a fadein / fadeout amplitude envelope. The type of envelope can be set in the option menu.
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 fader.

unit
The fader values from 1/1 to 1/64 note set the duration of one fader unit.
Additionally there are the settings : 'static'  -  which stops step-sequencing and remains on the first fader - and 'bufsized' which calculates the correct unit-duration for the length of the audio 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).

res
Adjusts the resonance of the filter used within the freq area.

dens
The density of the resulting audio stream. A value of 1 or greater leads to constant output, while a value smaller than 1 results in gaps. Be aware that the env fader also influences the occurrence of gaps. If you really want constant output, set 'dens' to 2.0 and 'env' to 0.



options
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 button.
To import options formerly saved in a .sxb file, select 'import options'. This will not change parameter settings or audio buffer contents.

options : 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 behaviour.
'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 output.

options : speed fader range
Sets the speed fader range :
    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 selected audiobuffer.

options : pitch fader range
Sets the pitch fader range with octave or semitone resolution. The center setting of the pitch fader is always at the original pitch.

options : filter cutoff frequency
Sets the 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 kHz.

options : envelope type
Sets the 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 to minimum results in the same window like the classic 'hanning' window, which can be selected, too, and where the env fader has no effect.
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.

options : playback mode
In 'loop mode' the audio playback will continously go on, while in the other modes playback will end when the cursor reaches either the audiobuffer end 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 position, that can be set with the small faders below each buffer . According to the triggermode setting, playback will stop or simply restart.

options : morph algorithm
The morph algorithm, that is being called when switching between two audiobuffers, can be selected here. 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 morphing.

options : morph stepsize
Determines the number of morphing steps between the two audiobuffers.

options : scope view mode
The type of audio visualisation in the small window at the bottom of splinetime can be 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.

options : record sync unit
When recording audio by pressing recsync , the currently selected audiobuffer will be erased and resized according to the selected sync unit value in this menu.

loadbank
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 menu. 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.

savebank
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 saved, too, but without the pure audio data. Instead the wav file references are stored, so that everything gets restored when reloading the song.

triggermode
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.

grid
Displays a graphic quantisation grid for the main faders.



mon
With this button switched on, the incoming audio of the mixer channel that reaches splinetime is being monitored.

recsync
Starts recording the incoming audio of the mixer channel as soon as the host-sequencer playback reaches the next bar. The audiobuffer length can be 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 .

recfree
Permanent recording of the incoming audio of the mixer channel into the currently active audiobuffer. Monitoring whats currently coming in is achieved by clicking mon . Stop recording by pressing 'recfree' again.
Use 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 always at 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 overwritten.

loadwav
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.

savewav
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.

killbuffer
Erases the currently selected audiobuffer.

amp
Sets the output volume.