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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, like us, consider a sampler to be a musical instrument in its own right, and not just a way to emulate other instruments. It has been a very high priority to make sure that adding and editing individual samples is as fast and logical as possible.

Well the real change I'm planning would not be so much about letting it die as it becoming freeware. I just don't feel good about it staying as a commercial product at the current rate it's developing. With a larger vibrant userbase I might just as well be more motivated to work at it than at the moment even if it's only role commercially would be as a surge-ad, who knows?

Sample/Intrument import

• 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)
• Propellerhead Recycle 1 & 2

Sampler engine

• 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)
• playback modes:
  

  forward
  forward loop
  forward loop with crossfading
  forward loop until release
  forward loop bidirectional
  forward shot
  sliced (maps slices accross the keyboard)
  on release
  reverse
  reverse shot

• 2 filters / voice
• 3 stepLFOs / voice. Doubles as 32-step stepsequencer 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)

keFIR is a VST, zero latency, mono FIR filter based on impulse responses loaded from wave files. Filter length can be adjusted to maximum of 8K taps.

It has a "mix" knob for mixing input with output, "gain" knob for setting output volume level, "length" knob for setting filter length (you can also set the length by clicking right mouse button on impulse response). Adjusting filter length will linearly affect CPU usage, more taps used require more CPU power. User can optimize it by choosing proper filter length, for example some amp cabinet impulse responses have long tail of 'almost zero' coefficients which can be shortened without audible difference, to save some extra CPU cycles.

Download @ Habib

MultiInspectorFree is a 31 band spectral analyzer designed to be used in a multitrack environment. Based on the technology of FaderWorks to handle multiple dependent instances of the same plug-in, MultiInspectorFree provides an easy way for simultaneous spectral analysis of several audio signals.
The first instance of MultiInspectorFree behaves like a standard 31 band spectral analyzer. Additional instances automatically send their spectral analysis result to all other instances of MultiInspectorFree. Every editor window of MultiInspectorFree shows the same content. Therefore, after opening all necessary plug-in instances it is sufficient to leave just one editor window open. The width and the color of the level bars are adjusted according to the current number of instances.

MultiInspectorFree supports up to 4 instances at a sample rate of 44.1 kHz.
MultiInspectorFree is available as a plug-in for Windows (VST only) and Mac OS X (VST and AU, both Universal Binary).

Download @ VertexDSP

Conventional compressors are based on peak or rms-level estimation to compute their time-variant gain or attenuation. This property of most compressors often causes undesirable intermodulation distortion and pumping/breathing artefacts.

JB PC-2 features a 'psycho-acoustic relevance' mode. Instead of using a peak or rms-level estimation, this mode employes a perceptual loudness model to compute the loudness of the input signal. This perceptual loudness model is combined with advanced attack and release stages that model peripheral adaptation of the human auditory nerve. The result is a very transparent compression characteristic, even with very short attack and release times.

On the other hand, the simple, classic compression behavior can be engaged too by simply switching off the complex loudness model.

Download @ Jeroen Breebaart