IO Configuration
The MiRA:: app supports up to 24 input channels. There are two main usages for this audio stream:
- Feed the RTA system; also called input reference (red parameters & information)
- Feed the capture system; also called TF input (green parameters & information)
The capture system is only available in MiRA::Live version
Although the audio channels are shared by both, it is of major importance to make sure that both systems are independent. Thus, one IO modification made on one does not affect the other one (as long as it is not common setting, such as the choice of audio interface, sample rate, etc.).
Source
Audio source
Audio source allows you to select which source to use as input. Depending on your current configuration and settings, this will include:
- Available SampleGrabber instance(s), either local or remote.
- Available hardware device(s), if one or several sound cards are present on the host system, and the corresponding device has been selected in the Hardware IO configuration dialog.
Hardware devices
Input & output devices
The FLUX:: MiRA:: allows for different input and output devices. The following sections describe the available options.
This setting lets you choose amongst a selection of devices, depending on your particular hardware configuration.
None
This turns off hardware input and output altogether. This is the recommended choice if you do not want to take advantage of FLUX:: Analyzer’s built-in audio capabilities, for example if you’re working with a SampleGrabber inside a DAW or Avid Venue console setup. With some sound cards that aren’t multi-client capable - meaning only one program can access it at once - disabling I/O is necessary to continue using another program simultaneously.
Your soundcard
Any installed soundcard(s) will be listed here. Under Windows, it might appear several times, in which case be sure to select the native ASIO driver for performance, not an emulated driver which be labeled something like ASIO DirectX Full Duplex Driver, Generic Low Latency ASIO Driver or similar.
External sampling rate
Allows the Flux:: MiRA:: to follow the sample rate settings of the attached audio interface.
Sampling rate
Sets the sampling rate used internally by the application. When a hardware device is selected, be sure to match this to the sampling rate set in the application panel of your soundcard control panel. We deliberately chose not to employ resampling, which, in our opinion, has no place in a measurement instrument. Instead, we generally advise you to set your soundcard’s sampling rate to 44.1k or 48k, which covers the entire audio hearing range (20-20kHz). Increasing the sampling rate above these values increases the processing power required to carry out the computations without any benefit for most practical applications.
Buffer size
Displays the current soundcard I/O buffer size. Depending on your soundcard, you might be able to change this to a different value directly in FLUX:: MiRA:: without opening its control panel beforehand. Smaller buffer sizes lead to a shorter latency between incoming audio, display updates, and audio output. This setting is certainly not as crucial as in the context of live sound processing, so there is no need to go down to extremely small values here, as this only increases the system load without offering any practical advantage.
Keep in mind a display refresh rate of 60Hz means one frame lasts for approx. 16ms, which is a bit longer than one 512 buffer at 44.1kHz, so the display will always lag less than one frame after the audio with such a setting.
Input (Reference)
Number of channels
Selects the maximum number of channels to be used by the application, or equivalently the number of channels in the application I/O bus. You should set this according to the source material format you want to analyze and visualize. This determines notably how many real-time curves are displayed in the Spectrum analyzer Presentation view, whether the Surround scope Usage is displayed, etc.
The FLUX:: MiRA:: supports up to 24 channels of audio.
Channel layout
Choose the channel layout for the reference input stream. New channel layouts can be imported by using the menu File>Import IO setup. Note that only .flux_io files are supported at the moment. All imported IO configurations are stored in the FLUX:: IO config folder, located at /Users/user_name/Library/Application Support/FLUX/IOConfigs on macOS and at C:\User\/...\AppData\Local\FLUX\IOConfigs on Windows.
Live (system tuning)
Max number of mic. / transfer function
Defines how many input channels are fed to the capture engine.
Channel specification
The channel specification table allows to specify options per channel:
- The name of the channel - display only
- The position (Azimuth, Elevation) - display only
- The routing (regarding the input and output device) - display only
- The name in the transfer function window
- Define whether the channel is used as a reference or a microphone channel in the transfer function window.
- Phase inversion
This table is shared between the real-time mode and the transfer function mode of the application. Information relative to the input reference is independent from the information relative to the transfer function measurement tools and the other way around.
Signal generator
Output
Selects one or two physical channels to which the Signal generator output should be sent.
In the case of stereo output, the signal is identical on both channels. This is provided as a facility for soundcards with minimal routing capabilities, and to avoid using a Y patch cable.
Feed input reference
Best practices in terms of function transfer measurement advice include making a physical loop between an output and an input of the audio interface to feed the noise generator into the reference input of the MiRA::.
In many scenarios, it is much easier to create a “software” loop. Activating the feed input reference sends the output of the noise generator directly to the reference input without the need for a physical loop.
Beware that, while being a handy option, we recommend using a physical loop as often as possible. The main drawback of the “software” loopback is that the latency of the sound card skews the delay measurements.