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HoRNet Angle is a little handy tone shaper tool. It’s based on the concept of the “tilt eq” a special kind of equaliser designed to change the tonal balance of the track to which is applied.
A “tilt eq” does not provide frequency or Q control, it simply increases or decreases high and low frequencies around a fixed point, and in the case of HoRNet Angle is 650Hz, turning the “Angle” knob on the GUI towards right increases the right part of the spectrum (the high frequencies) while lowering the left part (low frequencies) of the same amount. This simple action allow to perform very gentle tonal shaping that can help you get the correct frequency balance of your mix with very little effort.
HoRNet Angle also adds the standard HoRNet goodies, we have added input and output auto gain based around the 0VU level, leaving them engaged may help you keep the level constant while you tweak the EQ. HoRNet Angle also has the “OpAmp” analog emulation derived directly from our AnalogStage plugin that adds harmonics and make the sound “thicker” when enabled.
The best possible sound quality is always guaranteed by the auto-oversample feature that internally adjusts the oversampling amount to give the plugin enough bandwidth to keep the aliasing at minimum with every sampling frequency.
Tilt EQ adjustable from -18dB to +18dB.
Input and output gain controls with auto-gain.
Input and output peak meters.
Analog emulation based on AnalogStage’s OpAmp model.
Bypass button to exclude the plugin processing.
Mac OS X (>=10.7) and Windows support.
32 and 64-bit compatibility both on Mac and Windows.
Audio Units VST2.4, VST3 and AAX format.
HoRNet StereoView is a correlation meter designed to help you visualise the stereo image of your [b]tracks[/b].
The design of the plugin is very straightforward and easy to understand since the interface is completely occupied by the “goniometer”: an X-Y graph that displays the intensity of the signal in the stereo field.
StereoView allows you to change between two different views: the default, that we call “fireworks” that shows every sample as a straight line starting from the center, and the one we call “abstract” that connect each sample in the stereo field to the next one creating a continuous “wire”.
Another useful feature of the interface is the ability to be rescaled at will so you can resize it to fill the right spot on your screen.
A goniometer like HoRNet StereoView can be a very handy tool to directly see if your track have any anti-phase issue that may create problems when your track is played in mono, something that happens very often with mobile music listening today.
Display of stereo position and level of each sample.
Old CRT display persistence simulation for easier visualisation.
Two display modes: “fireworks” and “abstract”.
GUI can be freely rescaled.
macOS (10.7 and later) and Windows (Vista and later) support.
32 and 64-bit on Windows, 64bit on Mac.
Audio Units VST2.4, VST3, AAX format.
HoRNet CompExp is a compressor expander based on the design of a compander chip designed for telephone lines and used in many electronic devices to reduce the noise.
The chip works with a fixed threshold (in our case is internally set to -18dBFS) when the “amount” knob is at a value greater than 1 the gain is increased by the specified factor if the signal is below the threshold, and decreased if above. If the chip is set to work as an expander, the contrary happens, the gain is increased if above the threshold and decreased if below.
The system is designed this way to permit dynamic compression on telephone lines and compressor and expander are used together in this configuration, but we found that these models sounds interesting by themselves.
The model of this chip is implemented in the HoRNet HDD1 where is used to keep the noise down in the delay feedback loop.
Internally the gain is compensated so that the output is always kept to around -18dB RMS
Compression and expansion
Fixed ratio and gain compensation
Attack and release times
Mac OS X (>=10.7) and Windows support.
64-bit compatibility both on Mac and Windows.
Audio Units VST2.4, VST3 and AAX format.
Abstract Crystal Pads is a collection of 30 abstract pads for your film score, ambient soundtrack or your fresh new age project! Each sound is royalty-free and can be used in commercial and non-commercial projects.
Features of the plugin version:
30 abstract pads.
Amplitude range controls.
3 voice modes: polyphonic, monophonic, and legato.
Also available as a Kontak/Decent Sampler instrument library and a Reason 12/+ ReFill.
What? Shortcircuit is open source? Well yes, but there are big big caveats. Right now there are no downloadable assets you can get to play music, and there is no runnable open source Shortcircuit plugin. We estimate this will be the case for much of 2021, and maybe indefinitely. You may never be able to run an open source Shortcircuit. Might just be too much work. But if you are dev you can help!
What was Shortcurcuit?
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.
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.
Shortcircuit was designed to sound great, all other concerns have been secondary. But don’t take our word for it, download ‘shortcircuit free’ and find out for yourself.
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.
Over 2021, after the succesful release of Surge 1.9, the Surge Synth team undertook an effort to rebuild Surge in JUCE and add a variety of features. This project resulted in the new version of Surge, called ‘Surge XT’.
Synthesis method: Subtractive hybrid
Each patch contain two scenes which are separate instances of the entire synthesis engine (except effects) that can be used to create layered or split patches.
Quick category-based patch browser with textual search and favorites
Runs on a variety of operating systems, plugin formats, and architectures
3 oscillators per scene, with 12 versatile oscillator algorithms: Classic, Modern, Wavetable, Window, Sine, FM2, FM3, String, Twist, Alias, S&H Noise and Audio Input.
Classic oscillator is a morphable pulse/saw/dual saw oscillator with a sub-oscillator and hard sync.
Modern oscillator is a mixable saw/pulse/triangle oscillator with an optional sub-oscillator mode for triangle (which can also change to sine or square) and hard sync.
FM2/FM3 oscillators consist of a carrier and 2 or 3 modulators and various options.
String oscillator uses two filtered waveguides to emulate plucked or bowed string sounds.
Twist oscillator is based on a very famous Eurorack macro oscillator, and it offers a myriad of synthesis options in a very simple and quick way.
Most algorithms (except FM2, FM3, Twist and Audio Input) offer up to 16-voice unison at the oscillator level.
Oscillator FM in 3 different configurations and ring modulation between oscillators 1-2 and 2-3.
Most oscillator algorithms (except FM2, FM3, Sine and Alias) are strictly band-limited, yet still cover the entire audible spectrum, delivering a clear, punchy and clean sound.
Noise generator with variable spectrum (available directly in the oscillator mixer).
Two filter units in 8 different configurations.
Feedback loop available in 7 of those configurations.
Available filter algorithms: Lowpass (12 and 24 dB/oct, each with 3 variations), Legacy Ladder Lowpass (6-24 dB/oct), Vintage Ladder Lowpass (2 variations, each with and without gain compensation), Highpass (12 and 24 dB/oct, each with 3 variations), Bandpass (12 and 24 dB/oct, each with 3 variations), Notch (12 and 24 dB/oct, each with 2 variations), Allpass (3 variations), Comb (positive and negative), S&H.
Open-source additions to our filter algorithms include: K35 and Diode Ladder filter types from Odin 2 synthesizer, 12 and 24 dB/oct multimode filters from OB-Xd, and weird but fascinating Cutoff Warp, Resonance Warp and Tri-Pole filters by Jatin Chowdhury!
Filters can self-oscillate (with excitation) and respond amazingly fast to cutoff frequency changes.
Waveshaper (43 shapes).
12 LFO units available, 6 are per voice and 6 are global for the whole scene.
DAHDSR envelope generators on every LFO.
LFO shapes consist of 7 deformable LFO waveforms, step sequencer, a fully-fledged multi-segment envelope generator (MSEG) with various curves and up to 128 nodes, and last but not the least – a formula modulator which utilizes Lua scripting language to provide completely custom modulator output.
Voice LFOs allow envelope retriggering in step sequencer mode.
Extremely fast and flexible modulation routing. Almost every continuous parameter can be modulated!
16 effect units arranged as 4 inserts per scene, 4 sends and 4 master effects.
Iconic Vari-Mu limiting amplifier with extra features!
– VOLUME: -30 to +30dB input volume to drive circuit. You can drive circuit and get saturation plus more compression or vice versa.
– ATTEN: Output volume to adjust final level after process.
– THRESHOLD: Will let you set threshold to make compressor works. If you use VOLUME knob to boost input, you can leave THRESHOLD at default position and get mojo slightly compression. It has 30dB range.
– RELEASE: 300ms to 1.5sec release time
– HPF: 20 to 500Hz HPF to keep low frequencies from sidechain circuit. So, compressor won’t touch to low-end.
– EXT-SC: This will let you use external signal to compressed input signal.
– POWER: Plugin Bypass (Big Red Light)
– ANALOG OBSESSION: Oversampling (Label)
*Fixed Attack Time: 10ms
*Removed MIX option. Will be added with next update.
VST/VST3/AU | Mac 10.9 – 11.X
VST/VST3 | Windows 7 – 8 – 10
– Improved DSP and circuit design
– Added 4x oversampling
– Readout for knobs
– Redesigned GUI
– Catalina & Big Sur(Intel/M1) support
Pro Deluxe is here and it is free at the moment!
take full control of your sound and dive into the many
modulation options, effects and parameters with the
ability to layer up to 5 different waveforms.
Ability to layer up to five different waveforms.
2 x Oscillators.
8 powerful FX modules to bring your sound to life.
Easy modulation possibilities.
Unique Unison engine with 8 voices per oscillator.
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