Impulse Responses Stereo

Detailed: LSP Impulse Responses Stereo (IA1S)

Formats: CLAP, JACK, LV2, VST2, VST3

Categories: Reverb

Developer: Vladimir Sadovnikov

Description: 

This plugin performs highly optimized real time zero-latency convolution to the input signal. It can be used as a cabinet emulator, some sort of equalizer or as a reverb simulation plugin. All what is needed is audio file with impulse response taken from the linear system (cabinet, equalizer or hall/room).

Controls:

• Bypass - bypass switch, when turned on (led indicator is shining), the plugin bypasses signal (but still performs processing).
• FFT frame - the maximum size of the FFT (Fast Fourier Transform) frame that can be used for time-continuous convolution
• File - file selector, allows to load additional file that can be taken as impulse response for one of audio channels.
• IR equalizer - enables wet (processed) signal equalization in additional Wet Signal Equalization section
• Show - Displays the additional Wet Signal Equalization section in the UI
• Reverse - allows to reverse impulse file in time domain.
• Head cut - cut amount of milliseconds from the beginning of the impulse files, can be used to remove early reflections of reverb.
• Tail cut - cut amount of milliseconds from the end of the impulse files, can be used to remove large reverberation tail.
• Fade in - adds additional fading at the beginning of the impulse file.
• Fade out - adds additional fading at the end of the impulse file.
• Listen - this button allows to listen contents of the audio file.
• Source - this combo allows to select file channel to use for the convolution.
• Active - led that indicates that convolution is applied to the channel.
• Pre-delay - amount of pre-delay added to the processed signal. Can be used to individually control pre-delay of processed signal for each channel in the stereo pair that can provide additional stereo effect for reverbs.
• Makeup - amount of gain added to the processed signal. Can be used to individually control amount of processed signal for each channel in the stereo pair.
• Dry - amount of gain applied to the dry (unprocessed) signal.
• Wet - amount of gain additionally applied to the wet (processed) signal.
• Dry/Wet - the balance between unprocessed (Dry) signal and mixed signal (see Dry and Wet controls).
• Output - amount of gain additionally applied to the output signal.

'Wet Signal Equalization' section - visible only when IR equalizer parameter is turned on:

• Low-cut - sets the slope of the high-pass butterworth filter, possible slopes are 6, 12 and 18 dB/octave.
• Low-cut freq - the cutoff frequency of the high-pass butterworth filter.
• Faders - faders that allow to change the loudness of eight corresponding frequency bands in range of -12..+12 dB
• High-cut - sets the slope of the low-pass butterworth filter, possible slopes are 6, 12 and 18 dB/octave.
• High-cut freq - the cutoff frequency of the low-pass butterworth filter.