|Type / Tags|
Shortcircuit was created as a reaction against the ongoing trend where software samplers are being designed with the primary intent of library playback. It is intended for people who consider a sampler to be a musical instrument in its own right, and not just a way to emulate other instruments. High priority was given to ensuring that adding and editing individual samples is as fast and logical as possible.
The sample hierarchy in shortcircuit allow you to place samples directly at the highest level of the multi, without having to deal with instrument hierarchies and patches. Want to add a sample to your song? Just drag & drop the sample into shortcircuit and you're ready to go. Samples can be put in groups for multi-sampling and kit-building, but the complexity is only there when you need it.
Sound quality is of highest concern, and shortcircuit uses very high-quality interpolation to ensure that your samples sound as good as the source material, regardless of the pitch you play them at. All filters & effects are calculated at the precision required for them to sound the way intended and oversampling are used when required to prevent aliasing.
Each voice in shortcircuit features two filter-slots, and the selection isn't limited to the traditional pick. In addition to the regular lowpass / highpass / bandpass / notch & peak-filters and variations thereof there is an array of filter algorithms (not strictly filters in the traditional sense, but called so because of their location in the audio path) including distortion, parametric / graphic / morphing equalizers, bit-reduction / decimation, gating, limiting, slew-rate distortion, ring-modulation, frequency shifting and phase-modulation (better known as FM). The selection even includes analog-style oscillators that you can mix with the sample.
- Streamlined user interface for fast editing at the sample-zone level.
- Fast editing of multiple zones.
- "In context"-sample preview.
- Extensive drag & drop support (onto the keyrange-view or the list-view).
- RIFF wave-files (.wav) (8/16/24/32-bit & 32-bit float, mono/stereo at any sample rate).
- AKAI S5000/S6000/Z4/Z8 .akp banks (partial).
- NI battery kits (partial).
- Soundfont 2.00 (partial).
- Propellerheads Recycle 1 & 2.
- High-quality sinc interpolation.
- Oversampling used when needed to prevent aliasing.
- Double-precision float math (64-bit) used where it matters (IIR-filters).
- Single-precision float math (32-bit) used elsewhere.
- Supports any sample-rate.
- Max polyphony per instance: 256 voices
- Multiple outputs (max 16 mono AND 8 stereo-pairs per instance).
- Supported sample-playback modes:
- Forward loop
- Forward loop with crossfading
- Forward loop until release
- Forward loop bidirectional
- Forward shot
- Sliced (maps slices across the keyboard)
- On release
- Reverse shot
- 2 filters / voice. Filter algorithms:
- Lowpass 2-pole (2 types)
- Lowpass 1/2/3/4-pole ladder-filter
- Lowpass 1/2/3/4-pole ladder-filter with saturation
- Highpass 2-pole
- Dual bandpass
- Dual peak
- Comb filter
- 2-band parametric EQ (2 types)
- Graphic EQ
- Mörder OD (overdrive)
- Microgate (does glitch/loop style effects when the gate is open)
- Ring modulation
- Phase modulation (equivalent to FM)
- Frequency shifting
- Pulse oscillator
- Pulse oscillator (with sync)
- Sawtooth oscillator (with 1-16 voices in unison)
- Sinus oscillator
- 3 step LFOs / voice. Doubles as 32-step step-sequencer and wavetable LFO.
- 2 AHDSR envelopes / voice
- Powerful modulation system with the ability to modulate itself. Destinations include envelope-times, loop-points in addition to traditional destinations.
- Group LFO
- Group modulation routing.
- Group effects (2 effects / group). Effect types:
- Digidelay (feedback, filtering & optional MIDI-sync)
- Freqshift delay
- Freqshift flange
- Stereo width
- MS decoder