System Setup

The system setup UI scope lists all the inputs from the live system, along with their associated options. Here, you will be able to name the input, compensate for delay offsets, apply a target reference, associate a floor and a head microphone, etc. Each input gives four curves:

• The spectrum magnitude
• The phase response
• The coherence curve
• The impulse response.

Basic operations

The visibility of an input is defined by the toggle in the column.

To rename an input, double-click on its name in the ‘Mic./TF name’ column.

To change the color, double-click on the colored square in the column.

Gain & target

The column allows for adjustment of the gain of each capture, in decibels.

The button toggles phase inversion of the selected channel.

A reference target for the input can be defined in the column.A target is a previously recorded capture that is used as a reference for the input. For example, it is often used to calibrate measurement microphones.

Computation Curve

The computation curve appears automatically in the system setup list if more than one microphone is present. To feed an input into the computation curve, you must check its \(\sum\) column.

The computation curve has three different computing algorithms, which can be accessed in the “type” column.

• The averaging mode is recommended when using several microphones at different locations in the same venue. When using this algorithm, you can adjust the weight of a microphone by adjusting the \(\%\) column. One hundred percent indicates full contribution, while zero percent indicates no contribution. The coherence score also affects the computation curve: a measure with low coherence will have its weight lowered. You can deactivate this behavior by unchecking the combo box in the column.
• The sum mode simply adds the magnitudes of the different curves.
• The acoustic mode summarizes the magnitude, but it also takes phase relationships into account. This mode is recommended when dealing with separated measurements for heads and sub speakers.

Delay finder

FLUX:: MiRA:: uses an automatic delay-finding algorithm to determine the time-of-arrival difference between several microphones and the reference input.

The delay unit can be chosen from the drop-down menu. Possible options are:

• Delay in samples (smp).
• Distance in meters (m) or imperial feet (ft.).
• Delay in milliseconds (ms).

The delay finder always tracks for delay changes. The detected offset is displayed in the \(\Delta\) column in red.

Delay compensation

Pressing the button activates a delay line in the source signal path, compensating for the currently displayed delay value. This effectively aligns the source and response signals.

If necessary, you can manually adjust the delay figure using either of these methods:

• Direct keyboard numeric value entry as time or distance figure.
• Increment / decrement by clicking the +/- icons.
• Increment / decrement using the +/- numeric keys.

Delay adjustment considerations

Ensure stable conditions while performing a measurement

You should ensure both source and response signals have reached stability before attempting measurement. In particular, do not stop or start the audio, change the volume or any other parameter just before or during measurement. This would invalidate the measurement and you would have to start again.

Limitations

Please note there are many unknowns in play when determining the optimum delay figure. While we did our best to make this tool as robust and accurate as possible, there is always a possibility that it will fail, as with all automatic procedures. In this case, you should repeat the process or resort to manual adjustment until you get satisfactory results.

Multiple Paths

The major assumption behind delay compensation is that there is a main direct path from source to listener. This obviously does not apply in a very reverberant or complex-shaped acoustic space. This is where acoustic expertise and trial and error come into play in order to attain the best compromise.

Input Type

An input microphone can be of three different types. By default it is considered as a “Live (full band)” input.

The other accessible type is “Live Floor Mic (pair with..)”. When a microphone is switched to this type, it is expected to be placed on the floor to reduce the influence of floor reflection in the measurement process. It is then paired with another microphone. The one on the ground will produce data for the low frequency content of the measure, while the paired microphone will produce the data for the high frequency content. The crossover between the two microphones can be set in the configuration menus of the system setup.

The reference input displays a type of “Ref.” and cannot be edited.

The computation curve uses the type to define its averaging algorithm. See the section above.

Header buttons

For common scope header buttons, see the Audio Analysis Scopes section.

When the is engaged, it loops the first output channel of the signal generator into the reference input of the system setup. This setting is also accessible from the IO configuration menu.

The “Delay unit” drop-down allows for changing the delay unit, as seen in the delay finder section above. The buttons allows for changing the delay value of the select input of \(\pm\) 1 sample.

Options menu

The options menu is accessible by clicking on the button.

The Delay Finder FIFO parameter sets both the maximum findable delay time as well the global time averaging of the delay finder. It should be left to its default setting.

History time defines the length of the analysis buffer. It is set to 5 seconds by default.

Pairing crossover freq. sets the crossover point for floor/head microphone pairs.

Auto-pause: when the signal is below the level indicated on the Main IO, the Transfer Function will not be processed.

Columns visibility allows you to show/hide specific table settings.