Room Simulator

The Room Simulator is a reverb effect that simulates the reverberation of any room using its impulse response ("reverb tail"). A room impulse response is the reverberation of a very short, impulsive sound, such as a bang. You can additionally manipulate the impulse response with the various parameters in the dialog. This way you edit the reverb properties (reverberation time, damping...) similar to algorithmic, digital reverb effects. In addition, however, you have the option of fundamentally determining the reverb character by selecting the impulse response.

A graphic representation of the impulse response and the envelope allows an overview of the impulse response manipulation using the Room Simulator parameters dialog.

	Banks and presets: In the header of the dialog you can choose from a series of banks and corresponding presets. The default installation of Sequoia only installs a small selection of presets and impulse responses for the room simulator. You can get the rest of the presets and impulse responses by installing additional content via menu Help >Download instruments and sounds.... The presets contain both the parameter settings of the dialog and the reference to a specific impulse response. The impulse response can be any sample, supplied impulse responses have the file extension *.IMR.
	Impulse Response: File: Initially, the impulse response defined in the selected preset is selected here. From the menu you can select another impulse response from the folder of the loaded bank. To select the audio files opened in the program here as the impulse response, select the preset bank Loaded projects. At Info you get information about the length of the impulse response and whether it is in mono or stereo. With the button Play the impulse response is played. With Load any wave file can be loaded for use as impulse response.
	View dB lin./log.: The graphic display with the impulse response has linear amplitude scaling. This settings corresponds to the common display of samples. The representation in logarithmic amplitude scaling corresponds to the human loudness perception.
	Freehand envelope: The amplitude curve of the impulse response can be edited by drawing the yellow freehand envelope into the graphic with the mouse. You can use this to attenuate or cancel individual early reflections of the impulse response. With the zoom controls (Min, Max, +, -) you can zoom into the time course of the impulse response to find problematic reflections and edit them exactly. By clicking on the Reset Freehand Envelope button you can reset the yellow envelope curve.
	Pre-Delay: The Pre-Delay value delays the entire impulse response by a period of 1ms to 100 ms.
	Convert I. R. to Mono: If you select the Convert I.A. to Mono option, stereo impulse responses will be converted to mono. The convolution is calculated in stereo - in the surround case in n channels corresponding to the number of group channels.
	Calculate mono: If active, the entire reverb is calculated in mono only, which reduces the required computing power. The input signal and the impulse response, if stereo, are converted to mono before convolution.For surround application, all channels of the group are summed to a mono signal.
	Envelope I.R./Equalizer: Use these controls to influence the duration and sound of the reverb by editing the impulse response. Envelope I.A.: The envelope of the impulse response (light blue curve in the graphic) allows to fade in and out the impulse response. The edits made by the envelope are shown in the graph of the impulse response (red curve). By fading in the impulse response you influence the proportion of early reflections in the reverb signal, Env. Time defines the length of the early reflections, Early Refl. defines how much of it is faded in. At Late rev. you can fade out the reverberation by fading out the end of the impulse response. With Length you can reduce the duration of the Hall effect by shortening the impulse response. This will be cut off, so that an unnatural decay can occur, therefore combine the shortening with the fading by the parameter Late Rev.. Equalizer: Low: Using this parameter you can adjust the low frequency component of the reverb. High: Using this parameter, you can adjust the high frequency component of the reverb. FFT-EQ: The reverb signal can be post-processed with an additional FFT filter.
	Mix/Out: With these sliders you adjust the mixing ratio of the Dry and Wet signals as well as with Out the overall level. Wet/I/O: The level meters show the generated reverb signal (Wet) as well as the input and output signal level. Reset resets the markings of the highest levels of the level indicators.
	Parameter Presets: Here you can select, save, load and delete presets, which are composed of the parameters for the envelopes, frequency response and mix. The presets can thus be combined with different impulse responses.
	Performance/Options: Room simulation by convolution is relatively computationally expensive. With the settings at Performance Options you can set different qualities for the effect with different levels of computing effort. Quality: In the two modes Normal and Normal Advanced the room simulation is only calculated with half the sampling rate. In most cases this will be quite enough, since natural impulse responses and even impulse responses generated by digital reverb devices rarely posses components above 10 kHz (you can check this in the spectral representation of the integrated FFT Filter). The two Normal modes differ in the quality of the resampling applied, with the resampling quality applied in the "Normal/Advanced" mode being higher. The required calculation performance is slightly raised in this mode. In the High mode, the entire frequency scale will be calculated. The CPU load doubles in comparison to the Normal mode. Block length: This parameter determines the block length used for calculation. Short block lengths raises the number of the required calculation operations causing CPU load to increase. Long blocks lead to irregular CPU loads. The parameter has no effect on the calculation results. The Opt. Latency settings adjusts the internal block length in such a way as to keep the latency as low as possible without influencing the performance too much. The Opt. performance setting sets the internal block length so that the performance is as good as possible without letting the latency become too large. Note: It doesn't make sense to set a value smaller than the set ASIO buffer size. If the value corresponds to the ASIO buffer size, the room simulator processing is latency-free with the "high quality" setting. Get/Set Settings: This allows you to save and load quality options. When you have found a fitting setting, click on Set. This saves the setting. If you want to reload the saved setting, click the button Get. Bypass: This allows you to temporarily disable the effect in order to compare the unprocessed signal with the processed signal.

Tips and tricks

By applying effects to the impulse response you can influence the reverb character in many ways beyond the possibilities offered by the room simulator:
- If you apply menu Effects > Sample manipulation > Reverse to an impulse response, you get a reverse reverb.
- The application of time-stretching to the impulse response allows the room size to be changed without altering the resonant behavior of the room.
Apply an pulse-like decaying envelope to any shorter samples and use the result as an impulse response - this creates the most exotic reverb effects.
Interesting reverb sounds are also created when you use samples of percussion instruments as impulse responses.
Via Menu Object > Impulse response extraction… you open an assistant that supports you in creating your own impulse responses for use in the room simulator.