Impulse Responses Stereo
Detailed: LSP Impulse Responses Stereo (IA1S)
Formats: LV2, VST2, JACK
Developer: Vladimir Sadovnikov
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).
- 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
- 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.
- 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.